Re: [asterisk-users] Writing CDR's to two database servers
On 19/6/17 4:47 pm, Tech Support wrote: I know that there are probably several solutions to this problem, but what I am trying to do is provide some redundancy for my customers CDR data. I know that doing simple backups of MySQL is probably the easiest way to go, but I'm thinking that there may be some benefit to simultaneously writing the CDR data to multiple servers at once. However, I'm drawing a blank on this one. Has anyone else done this before? Any insight at all would be greatly appreciated. You could - if you really wanted - use two different cdr_ modules to write to, for example, a MySQL and a PostgreSQL database simultaneously. Having said that, and given nearly every modern DBMS has its own replication built-in, you'd be far better off using that. There are good instructions online for MySQL and PostgreSQL - and no doubt for other DBMSs as well. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call List Campaign to an IVR
On 6/2/17 5:24 pm, Tech Support wrote: Basically, two calls are made. ... When the first call is made for such a short period, the remote end still goes off hook, but the call will end before it starts to ring. Then, halfway through the first call, a second call is made. Since the remote end is off hook from the first call, the second call will get sent to voicemail and the message is played there. Am I right in thinking call waiting isn't a thing on US mobile networks then? In the UK, call waiting is pretty standard, and almost universally enabled by default on mobile networks. AIUI the same is true for much of Europe. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone provisioning template Snoms
On 7 May 2015, at 23:45, Tafadzwa Nyabasa tnyab...@gmail.com wrote: I am looking for a phone provisioning template for Snom phones, Yealinks and Polycoms. I am always doing deployments of many phones and usually configure each phone one by one for each installation. Any help will be highly appreciated There’s some excellent documentation about provisioning on the Snom Wiki: http://wiki.snom.com/Category:Auto_Provisioning:Configuration_Files You can set the phones (via DHCP options) a firmware url on a web server under your control, grab their MAC addresses, then deliver them custom config settings as required. Easiest way to start is to copy the config file (via the web interface) from a phone with factory default settings, then just change the settings you need to change, and write something in your scripting language of choice (PHP, Perl, Python, etc.) to just send those settings to the phone dependent on MAC address. Don’t send *every* available config setting to the phone - only the changes from default you need to make. I suspect the same can be done with Yealink and Polycom phones - I’ve not used those so can’t really comment. I have a similar system which seems to work for Sipura/Linksys/Cisco phones, though most of my new deployments are exclusively Snom. Kind regards, Chris -- C.M. Bagnall This email is made from 100% recycled electrons -- _ -- 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] Anonymous SIP calls
On 27/3/15 8:03 pm, James B. Byrne wrote: One only accepts VOIP calls from known correspondents. I am not clear why this is so other than vague warnings respecting (admittedly real and serious) security issues. Because on the whole most people don't *want* to receive calls from random strangers :-) What is it about incoming SIP calls destined to our internal users that make those calls so dangerous? Why cannot incoming anonymous SIP calls not be treated exactly as incoming PSTN calls Others have already written far more eloquently than I about the security implications, but I think there are other factors at play here. One of the principal benefits E.164 brought to the table was the ability to 'bypass' the telco (and their call charges) and route the call direct to the desired endpoint over our respective internet connections. But the cost of making calls via the PSTN has reduced to a point where the cost of the call is no longer a significant factor in whether to place the call. Think back even a few years: the cost of calling another country could easily rise above 1 (GBP/USD/whatever) per minute. Now, with the exception of a few far-flung locations, there are very few destinations to which calls are even a fifth of that cost. Calls that come via the PSTN are subject to some sort of regulation. Bonafide marketing companies are obliged to screen their calls through the TPS (in the UK - I presume there's a similar 'do not call' screening process in other countries). It's not perfect (international marketers aren't effectively covered, for example), but it is marginally better than a total free for all. As for solutions, I think that for direct SIP-to-SIP calling to gain the traction originally promised, we need to get to the same level of incoming call control as we have with spam filtering on email. So there will need to be organisations running distributed RBLs similar to (for example) Spamhaus which SIP servers can query in real time to check not just for hack attempts, but also those SIP servers from which unsolicited marketing calls have originated, etc. In summary: 1) PSTN calls are now /cheap enough/ that the financial benefits of direct SIP-to-SIP calls for most users are negligible. 2) When the cost of calls falls to (effectively) zero, the principal beneficiaries are fraudsters and telemarketers, and most people would rather not deal with either group. 3) Lack of effective protection - both technical and regulatory - against SIP-to-SIP misuse (not just fraud, but unsolicited callers, etc.) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] BlindXfer Sensitivity
On 16/2/15 4:13 pm, Andrew Colin wrote: The strange thing is its only sometimes my dial string is as follows exten = s,1, Dial (SIP/200,, tT) For that particular route but obviously s,3 as have Ringing () first etc. After she pushes ## 6 times it will go thru sometimes. Are you sure it's a DTMF sensitivity problem rather than a delay problem? I've had several sites where the default DTMF timeout of 0.5 seconds is too short for users to achieve, and have set featuredigittimeout (in features.conf) to 3 seconds to give them more time to press the combinations they need to press. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
On 29/10/14 12:59 pm, A J Stiles wrote: Imagine what would have happened to the human race if Ugg the Caveman decided not to share the secret of making fire with everyone freely, but instead went around demanding shiny beads with menaces from anyone who just wanted to keep themselves warm . That's the best analogy I've heard in favour of open development for a long time, and something that non-techs can understand. I thank you sir :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] PBX hacked: why hundred of calls to the same number ?
On 3/10/14 6:52 pm, Rainer Piper wrote: the attacking server changed the destination Number at 18:53 CEST and he is still blocked ... LOL 972597438354 callto:00972597438354 It's pretty much an everyday occurrence for any internet-connected SIP system these days... Oct 3 19:46:20 server /sbin/kamailio[3977]: NOTICE: script: blocking IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=100972597438354 Many of these attacks come from fairly easily recognised user-agent strings, so if you fancy doing a bit of packet inspection with your firewall, you can block many of these before they get as far as your SIP server(s) themselves. For example, the sipcli scans you listed above can be blocked fairly easily with: iptables -A INPUT -p udp --dport 5060 -m string --algo bm --string sipcli -j DROP (obviously there are overheads to string searching UDP/5060 packets that you'll want to consider, and the above won't work if you're using sipcli legitimately anywhere on your network) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to strip +1 out of incoming number
On 2/10/14 6:52 pm, motty cruz wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
${CALLERID(num):1} should do what you're after (or :2 if you need to
strip the + as well)
Kind regards,
Chris
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Re: [asterisk-users] Playback/background audio from MySQL BLOB
On 24/9/14 10:36 am, A J Stiles wrote: But personally, I'd just store the filenames in the database; and rely on the unix filesystem for storing the actual file contents. After all, that's what a filesystem is for. This. Shocking as it might appear, filesystems are remarkably good at storing files. They were designed to do it. Why try to shoehorn a database into doing something it wasn't designed to do (and isn't particularly good at doing)? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 secure fine tune - stop attack
On 4/9/14 4:58 pm, Eric Wieling wrote: If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses by using iptables. This takes care of at least 80% of attacks. Likewise here (though RIPE rather than ARIN, since we're the other side of the pond). You can also take it a bit further: if, for example, you know what ISP(s) your dynamic clients are using, you can limit connections to the IP ranges those ISP(s) use - look up their ranges on he.net's BGP looking glass if you need to find out what ranges they're using. Another thing I've been playing with of late is using iptables' string matching functionality to block user agents of known attack vectors: 'sipcli', 'sipvicious', 'friendly-scanner', etc. This seems to work remarkably well, though what impact it has on net performance under load remains to be seen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Internal timing under load is critical ?
On 30/7/14 10:08 am, babak wrote: I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely struggling to think of (legal) reasons why you'd want to broadcast 1000+ simultaneous calls. Perhaps I'm just not being imaginative enough... :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Limit Asterisk
On 23/7/14 10:29 pm, Steve Edwards wrote: Don't buy hardware until you've identified (either empirical or calculated) the bottleneck. If you've plenty of spare RAM (and at 16GB I'd suggest you probably do), I'd throw in the possibility of recording to RAM disk, then moving the calls to hard disk during your quiet (or closed) hours. SSDs do rock. I recently observed (via vmstat 5) a Samsung 840 topping out at 460,000 blocks per second. I can remember when 10,000 was big :) This. The 840 is a great bit of kit - we've replaced nearly all our spinning disks with a mix of Samsung 830 and 840 drives. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Voicemail message to text
On 20 May 2014, at 15:35, Ishfaq Malik i...@pack-net.co.uk wrote: I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? With the huge variety of different accents and intonations in human speech (even in one country), my experience of all speech-to-text engines has been one of poor accuracy at best. If you need messages-to-text, generally best to use a virtual PA company or similar - at least in my experience. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] cdr viewer for csv
On 24 Apr 2014, at 11:36, binary dreamer dreamer.bin...@gmail.com wrote: I am running asterisk and all of my CDRs are in the default csv. the system is so limited to ram (only 256) and I cannot run MySQL or any other program to give CDRs a fancy view. As an aside, have you considered running your CDR storage/viewing on a separate machine? You don't have to log CDRs on the same box as you run asterisk. at the moment the only other software running is nginx for a static webpage with guidance on the system. is there a way to present to a webpage the CDRs from the csv, please? You can almost certainly do this if you want using the standard string handling functions in $middleware_of_choice, but the lack of indexing on text files will make this *very* slow for search queries etc.. The RAM/CPU requirements associated with loading huge chunks of text data into memory, manipulating them, then displaying the results will likely exceed that of a DB. Unless you only want a recent call log, you really want to do this in a database. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Open Source Asterisk Polling Solution
On 22/4/14 11:44 am, A J Stiles wrote: Firstly, be warned: Are you sure that is even legal to do in your jurisdiction? You could be setting yourself up for a hefty fine! Check applicable local laws before proceeding. This. I'm glad someone else thought it worth mentioning as well :-) Even if it's letter-of-the-law legal, consider whether you'd be willing to be on the receiving end of it. If it's providing a useful service, i.e. calling a specific list of individuals who might have medical appointments in the coming week to confirm they don't need to reschedule, then fair enough, but if you're phoning people who've not consented to sell them double glazing, then you aren't doing your client's reputation any favours and you're going to mightily annoy those you're calling. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Old Asterisk Release wanting to upgrade ...
On 17/4/14 3:53 am, Lee, John (Sydney) wrote: I have written a lot of AEL2 script in Asterisk 1.4.x and I am not sure if it will still run in 11. If I'm honest, this is why I still have so many 1.4.x boxes around as well. I've been using 11 for new installs, but the thought of having to redo all the AEL macros from 1.4 does not fill me with any enthusiasm to update those boxes. The switch to Gosub() does not seem to be an easy drop-in replacement for Macro(). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Live Recording on the Storage Server?
On 17 Apr 2014, at 16:14, Paul Belanger paul.belan...@polybeacon.com wrote: hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. +1 save yourself the headache and do this. I'll add another +1 to this. I've never been able to get multi-channel recording (even 3 or 4 channels) working reliably over an NFS link to another server. I suspect, with some tweaking of nfs options it might be possible, but if it ain't broke… Just a cautionary note if you do use a cron job to move recordings to a storage device at regular intervals: make sure you use lsof or similar to check the recordings aren't actually open by asterisk at the time, otherwise interesting things will happen. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Old Asterisk Release wanting to upgrade ...
On 17/4/14 4:53 pm, Eric Wieling wrote: I had little problem converting my AEL scripts from 1.4 to 11 Did they have lots of macros in them? If so, then you, sir, are a better man than I, and I take my hat off to you :-) (and any hints you might want to share in converting 1.4 AEL macros to 11 would be gratefully appreciated) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Replying to Posts
On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. On 13/3/14 5:13 pm, Ron Wheeler wrote: -1 Prefer top posting. Easy to see if I want to scroll down to see if it is something interesting to me. I get a lot of e-mails each day and scrolling wastes too much time. You can then also reply to another point here like this: it's as much about trimming previous posts as about not top posting. New posts should include just enough context to ensure the message isn't meaningless, but not quoting a 20+ line irrelevance. That way you see both the question and the answer without scrolling. Whilst we're on the subject of mailing lists, I'd like to add my personal pet rant: MTAs that don't add/honour In-Reply-To headers. Completely breaks threaded readers. That is all :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] High Availability with Asterisk
On 6/3/14 3:21 pm, Thorolf Godawa wrote: The idea would be having a HA-cluster of two servers with Xen, each of them runs one instance of an Asterisk-system in a single VM and on a failure the VM will be restarted on the other node. This might result in a much higher load on this node, because is runs two VMs, but for a short period, until the other node comes back again, it might be tolerable. This is basically what we do, though in our case we use KVM rather than Xen; we found KVM behaved a great deal better managing timing than Xen, but YMMV and Xen may well have come along a great deal since we last looked at it. In fact, it could be argued that even without any need for HA, there's still an advantage to running a server in a VM: hardware portability. If the machine dies, you can quickly redeploy the VM to a new host without having to recompile things and so on because hardware has changed. Are there other options running two Asterisk-instances parallel on one system, each binded on it's own IP, maybe s.th. with chroot or similar? You might be able to do something interesting with containers (LXC), but given the ease of setting up KVM and the (relatively) small performance overhead, we've tended to just stick with that. On 6/3/14 3:46 pm, Michelle Dupuis wrote: A lot of HA tools don't look deeper into Asterisk to see if/how it has failed (they only detected catastrophic failures). What happens when the Asterisk process is alive but no longer bridging calls? In fairness, the tools the OP mentioned (pacemaker/corosync) can be set up to detect other failures than whether asterisk is alive - a simple one to set up is to try connecting on 5060 UDP and make sure you get an acknowledgement. Likewise, you could even set up a call using the manager interface to a dummy extension and make sure it completes successfully. FWIW, we tend to use pacemaker with heartbeat rather than corosync, but both perform a pretty similar function. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] G729 Licensing Revisited - I'm Sorry!
On 28/2/14 9:04 pm, Jayson Devor wrote: That being said, will purchasing 23 licenses (one for each channel that we use), and continue to use the open source g729 sorftware keep us legal? I know at least half a dozen people who do this so that they can more effectively balance their licence commitment over a number of services, rather than locking licences down to MAC addresses of specific NICs in specific servers. But I'm based in the EU where (as others have said) patentability laws are quite different. If you're worried about whether it's legal in your jurisdiction, you really should speak to a qualified legal professional to allay your concerns. This list has such an international audience that what's perfectly acceptable in one jurisdiction might land you in hot water in another. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 NAT
On 19/2/14 4:53 am, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? I can't help on the can Asterisk detect they're behind the same NAT part of the question, but I would caution you that an assumption that 'each NAT box has a single external IP' is risky - especially if you have to deal with the possibility of double-NAT and other such evilness (and it's hard to avoid sometimes - how many non-tech people go and buy a wireless router to 'extend their WiFi' rather than an access point, or don't know how to switch said router into AP-only mode). You also have to consider users who have multiple LANs (which might not necessarily be able to route between themselves) behind a single external IP: this one's quite common in shared/managed office environments - one external IP and several RFC1918 VLANs internally, with no routing between them. So in summary, unless you have a considerable level of control over your endpoints such that you can be sure these (and no doubt other) scenarios don't apply, it's probably safest to send RTP traffic through Asterisk regardless, otherwise you're potentially opening up a support nightmare for yourself. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Want Queues to ignore mobile operators voice mails and continue ringing...?
On 14/2/14 9:21 am, Gareth Blades wrote: I would suggest using the 'M' option on the Dial command to run a macro. The macro can just wait fir a key to be pressed and until it is pressed the Dial is still effectively ringing. So if it does go to voicemail then the call wont get put through. You need to make sure you have a suitable value set to abandon the agent call if its ringing too long. The callee may also find they are left multiple voicemail messages. This is the approach we've used in the past: force the recipient to hit a button to accept the call, something which their mobile voicemail will never be able to do. The alternative - and it only really applies if you have control of the mobiles in question - is to disable the mobile network's voicemail service entirely, and manage diverts from the handset. That way you can then recreate your own 'mobile voicemail' service on your asterisk platform with all the normal asterisk VM benefits such as email delivery, etc. You can then of course detect when those mobiles 'divert' to voicemail (since it's now on your system), and kick them out of the queue at that point. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Want Queues to ignore mobile operators voice mails and continue ringing...?
On 14/2/14 10:54 am, Tiago Geada wrote: How does one detect the 'divert' to voicemail? If you're using the mobile network's voicemail service, you can't as a general rule; you've no reliable way of knowing whether that call was answered by the user or their voicemail service. However, if you're providing the mobile voicemail service yourself from your asterisk platform, then you can detect the *incoming* call from the mobile device in question to their mailbox and act accordingly. As I said in my earlier reply though, it depends on you having end-to-end control of the mobile devices in question and your mobile operator will allow their voicemail service to be completely disabled. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] IOPS required by Asterisk for Call Recording
On 25/1/14 5:26 am, Amit wrote: How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I suspect this might be your problem: 250GB SATA disk (No RAID) Is there any way to tune / optimize / configure for better write performance? Perhaps consider recording to a ramdisk first, then periodically write out completed files to HDD at your leisure (e.g. during slack periods)? Or, given the relatively low cost of 250GB SSDs these days, swap out the spinning disc for an SSD. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] stopping unwanted attempts
On 19/1/14 2:57 pm, Ron Wheeler wrote: fail2ban is so easy to set up, there is no reason not to set it up. One of the dangers with fail2ban - at least in its default configuration - is that a legitimate SIP phone with an incorrect password can quite easily send dozens of registration attempts in a couple of minutes, thus blocking that IP. If your end users configure their own phones, you will have to factor in the increased support burden when users complain that their phones 'can't connect' and you need to manually unblock those IPs. This can be at least partially mitigated using fail2ban's 'ignoreip' directive for IPs you know only your users will be connecting from. If you've a large number of users, it might be worth splitting them across a pair of servers - one for 'trusted' users, i.e. where each SIP endpoint is locked down to a specific IP (or at least a range), and you can configure your firewall to block SIP connection attempts from anything apart from that list; and one for 'untrusted' users, i.e. travelling users, home workers without static IPs, etc. on which you run fail2ban with a fairly ruthless set of rules/limits. Unless you know that none of your users travel internationally, I'd be wary of imposing countrywide IP blocks, especially in this era of IP shortage where IP space is being traded on the open market and GeoIP databases may not always keep up to date. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Text to Speech Engine
On 10/1/14 8:16 pm, Jai Rangi wrote: Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. We recently used Ivona for a fairly complex IVR project (multi-lingual, including pronunciation of foreign names). http://www.ivona.com Not free, but we found the sound quality considerably better than we were able to get from either Festival or Cepstral. Worth bearing in mind that we are based in the UK, so our primary concern was for good quality British English voices. I cannot comment on other variants such as Australian or American. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Mass exodus
On 13 Nov 2013, at 18:29, Mike Diehl mdiehlena...@gmail.com wrote: I've been seeing some strangeness lately on my 10.2.1 server. It's gotten to the point that a few times each day, I see masses of SIP clients becoming unreachable. They're not all on the same network, and we don't see any calls drop. In a few seconds, they all come back. I don't think it's a connectivity issue because we don't drop calls, and the endpoints aren't on the same networks. We don't see excessive CPU load when it happens. It does SEEM to happen most right after someone accesses their voicemail. We saw this happen on a 1.4 server a couple of years ago shortly after 2am each day. It was only after a study of the cron schedule we narrowed it down to a number of rsync backup jobs which were run at that time. As in your case, it wasn't a connectivity or bandwidth issue - in the end we put it down to a disk I/O bottleneck. It might be worth running something like iostat on your box to see if you see a spike in iowait as voicemail is being checked. We resolved it simply by rate limiting our rsync jobs. In your case with a busy database, you might want to look at your MySQL indexes and/or cache settings - this might be something worth asking about on the respective MySQL discussion groups as well as here. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] What linux distro most popular for Asterisk
Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)also hoping for something more current. Mix of Gentoo and Ubuntu here (Gentoo mostly on old Via Epia MiniITX systems which don't have full i686 instruction set support). The best Linux distro is usually the one you're most familiar with - that way, if/when something goes wrong, you stand a reasonable chance of being able to fix it. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Using sqlite3 for CDR logging
On 3/10/13 5:52 pm, Tech Support wrote: I was thinking of using sqlite3 to log CDR's, thinking that would be faster than using MySQL. Has anyone ever benchmarked this to quantify just how much faster sqlite3 is? Are there any drawbacks to using it? Lack of multi-user concurrency is the big one. At the risk of encouraging database contests on the list, have you tried using PostgreSQL instead? It's a gross generalisation, but In my experience, PG handles writes better than MySQL, which in turn tends to handle reads a little faster than PG - assuming both are in 'out of the box' (i.e. unoptimised) conditions. If you wanted to stick with MySQL, you might want to have a go at optimising it - there are quite a few scripts knocking around the web which run a set of queries on your data and suggest optimisations to apply. And others have said, running the DB on a separate host is never a bad thing, and ideally on SSDs or RAM storage if you can. Spinning disks are often the bottleneck with large data sets. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Dialplan MySQL inserted ID
On 20/8/13 5:00 pm, A J Stiles wrote:
Why not write an AGI script in your favourite language (Perl, Python, PHP,
Java all have AGI and MySQL bindings) to perform the INSERT query for you?
+1. It would also give you somewhere to perform sanity checks on your
${ARGS} to avoid SQL injection attacks...
Kind regards,
Chris
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Re: [asterisk-users] Paltel subscribers as called parties for SIP attacks
FWIW, we routinely see dodgy traffic from: ovh.net hetzner.de But since those are 2 of the larger short-term contract dedicated server vendors, I'm not surprised about that. It's so frequent that I don't even bother reporting it any more - when an abuse report is acted upon and the server shut down, another pops up to take its place. all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers). Likewise here. Well, not all, but a sizeable percentage of it. We're based in the UK. Why would an internet subscriber from hadara.ps, for instance, want to call a Paltel mobile user via some remotely hacked SIP PBX thousands of miles away given than Paltel is partially owned by Hadara Technology Investment Co. (and Paltel leases long-haul infrastructure from Hadara anyway)? Are you perhaps reading too much into it? There are insecure servers and computers all over the internet. These are (ab)used and co-opted into botnets which are in turn used to compromise SIP servers. I suspect that it's probably a financial goal (free calls, or substantial termination payouts) rather than a political goal the perpetrators are seeking. I'd be curious to know what everyone else's experiences have been like, and why 95% or better of the SIP attacks on my PBX are destined for Paltel mobile subscribers. Perhaps the termination payout on those numbers is particularly good, and/or regulation/investigation into abuse isn't so good? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] include directory with multiple files in it
On 5/8/13 2:18 pm, Jonas Kellens wrote: is it possible to use the #include - syntax to include several configuration files situated in one directory ? Something like : extensions.conf : #include extra/* #include addons/* Is this possible ? Yes. You can also do crafty things like: #include */extensions.conf Which will include the file extensions.conf from each subdirectory. Very handy if you have a structure like this: /etc/asterisk/client1 /etc/asterisk/client2 etc. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Dongle or extra channel and sip SMS
On 15/7/13 3:00 am, bilal ghayyad wrote: I need to be able to send SMS messages for campaign or for specific users, also I need to be able to receive SMS messages and do automatic reply. In my experience, SMS is something best done out of Asterisk. That's not to say that Asterisk can't do it, of course, just that there are providers out there who can give you a nice friendly API for easy integration into your application. This is especially true if you need to send *lots* of messages in a short space of time: simply adding a single mobile device with a single SIM isn't going to cut it - you're going to need a bunch of them, at least. All of those will likely have different numbers, so you're going to have to handle that for receiving messages. Then you have to consider that some networks will charge more to send messages to numbers on the same network vs. a different network, so you might have to separate out your numbers into networks (easy if they've never been ported; more tricky if they have). Based on past projects (in the UK), the cost of multiple SIM contracts, the necessary hardware to connect them, development time, etc., is usually more than the cost of paying a third party with a suitable API x per message to deliver them on your behalf. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] PoE module
On 14/7/13 8:12 pm, bilal ghayyad wrote: We have a cisco switches but they are not PoE and we need only to have PoE device so the cables come for it first to provide the power and then goes to the switch (to be like batch panel), is there something like this that can be used for the IP Phones? Have a chat with your usual network equipment supplier for midspan PoE units. Phihong make some, and those I've used seem to have been pretty reliable. There are no doubt many other suppliers of such things. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] PoE L3 Switches
On 14/7/13 11:45 pm, Gregory Malsack wrote: I've used a lot of Dlink DES-1228p and 1210-28p. Primarily with polycom phones. Seem to have pretty good luck with them for the last 7 years or so. +1. We've used quite a few DES-1228P units in the past, and apart from 2 early unit failures, we've not had any failures since (out of a few dozen). The management interface isn't great, especially if you're used to the command line goodness of an HP or Cisco unit, but provided you aren't fiddling with it too often, you'll manage. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] I need a second opinion on a new phone system deployment
On 15/6/13 7:00 pm, Carlos Alvarez wrote: Interesting product that I was very interested in, but the licensing has one huge glaring problem. Be sure to read the FAQ carefully. If your hardware fails and you replace almost anything in the machine, you have to pay for the product again. Not to mention that installing Pacemaker/Heartbeat/Corosync or your other HA solution of preference isn't particularly difficult, and is agreeably free. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 HA
On 6/6/13 4:53 am, Gopalakrishnan N wrote: Any other HA applications available or the lsyncd with pacemaker is good? I generally use Pacemaker with Heartbeat, which seems to work pretty well. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Issue in transcoding
On 2/6/13 2:01 pm, Muhammad Yousuf wrote: I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 Isn't g723.1 considered pretty poor quality these days? Can't you set voipswitch to use something apart from that? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Auto dialer scripts and software
On 22/5/13 10:54 am, A J Stiles wrote: You do know that sort of thing is against the law -- or at least requires a permit from the authorities -- in most civilised countries, right? And it's worth adding that even if it is legal in your country, you're almost guaranteed to offend/annoy your target audience. Recorded calls always do. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 Log rotate not working
On 21/5/13 4:19 pm, Ahmed Munir wrote: Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily basis which was working perfect. Now in couple of months back, the logrotate feature is not working at all but simply appending the logs in 'messages' file. Listing down down the configuration for logrotate below; This sounds more like a Linux/logrotate issue rather than asterisk-specific. Are your other system logfiles successfully rotating? (e.g. /var/log/messages) If not, it may be something as simple as logrotate's daemon not running. You should be able to fix that in your distro's startup scripts. On Gentoo, you'd do something like /etc/init.d/logrotate start to start it now, and rc-update add logrotate default to add it to your default runlevel. Difficult to advise further without knowing the distro in question. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Performance Asterisk large installation on Vmware/Xen
On 18/5/13 8:09 pm, Mitul Limbani wrote: Not recommended to run Asterisk on Virualization I used to share that view, but having done a few medium-sized installs recently in virtualised environments and encountering no problems to speak of, I'm not sure it's necessarily the case any more. Things to look into closely: - passing hardware devices from bare metal to VMs is at best 'imperfect', so if you need PSTN connectivity using ISDN or POTS cards, you're probably best doing that in physical hardware. IP-only stuff seems to be okay. - be *very* careful about the load on the host machine. If you have total control over the VMs running on each machine, you'll probably be okay, but if you have to share a bare metal host with other VMs over which you have no control over the load, you'll run into problems. - you may come across problems with timing sources for conferences and the like, though I understand this has improved considerably in recent asterisk versions (i.e. no dependency on dahdi_dummy or ztdummy any longer). FWIW, I've recently been using KVM as an alternative to both Xen and VMware, and I'm very impressed. It's certainly my preferred VM platform at the moment (not just for asterisk stuff, but in general). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Monitoring SIP trunk status on call by call basis
On 14/5/13 4:30 pm, Ishfaq Malik wrote: I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my primary goes down. I'm wondering what the best method of checking if the primary being up is. Well, the obvious start point might be ChanIsAvail() - that'll at least weed out an upstream SIP peer that's unavailable (assuming you're using qualify) before you even get as far as Dial(). However, one of the problems you might encounter when sending calls to a provider is an inability to distinguish between Congestion and Busy. Ideally, of course, you want to route the call to upstream2 if you get Congestion from upstream1, but not if the dialled number is Busy. There's not always a good way around that. As others have said, the only real way around it is to send calls periodically to verify end to end operation - at least this way you're testing both your upstream's SIP connectivity and also their PSTN termination. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] looking for a way to do appointment reminders
On 26/4/13 10:38 am, jg wrote: they are currently calling patients. I think these calls apply only to a certain fraction of the patients, who are difficult to contact by other methods. I suspect there will be different requirements depending on how 'helpful' to patients you wish to be. At the very simplest end of the scale, you could simply call the patient's number and remind them of their appointment on dd hhmm, then disconnect. However, the OP probably wants something a little more sophisticated than that. At the very least, you would want some method of handling shared numbers (e.g. a shared dwelling with a single phone), so you didn't inadvertently advertise a patient's appointment to someone else who answered the phone. So you would at the very minimum want a simple IVR that says We are trying to reach Mr. Joe Bloggs. If this is he, press 1 now, otherwise please hang up. Going beyond that, you might want your reception staff, when booking appointments, to ask the patient when they would like their reminder call - the day before, an hour before, etc. etc. (and if the day before, would they prefer it in the morning, afternoon, or evening). As others have said, the OP might be best advised to request (paid) assistance with the project on the [asterisk-biz] list. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] looking for a way to do appointment reminders
On 26/4/13 10:14 am, Hans Witvliet wrote: Only reasonable option is to send them an SMS. Given the likelihood that a sizeable percentage of people attending a medical establishment are going to be at the upper end of the age scale, it's possible they may not have mobile phones, and even if they do, might not understand how to read SMS messages on their phone. Probably would work okay for certain establishments, but I'd be wary of exclusively using SMS in a medical context, given the likely patient demographic. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] looking for a way to do appointment reminders
On 26/4/13 12:24 pm, jg wrote: This way the callees would always talk to a human being If possible, this would definitely be a Good Thing. Many people (myself included) will disconnect a call as soon as they realise it's a recorded message. It also means the human caller can confirm they really are talking to the patient (perhaps by asking their DOB or similar). It may be possible to outsource something like this to a Virtual PA service or similar. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Network based transcoding
On 12/4/13 4:38 pm, Nick Khamis wrote: We were looking more into the lines of a scalable multi server router like a cisco 3745. Perhaps it might help to tell the list just how many concurrent calls you're looking to transcode? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Serviced Office operator panel
On 11/3/13 11:07 pm, Andrew Yager wrote: Basically, if you know of a product, open or closed source, and would like to sell it to me and you think it does the job, or you've seen something that works, contact me off list ASAP! Actually, please post *on* the list if you know or have used something that meets the above. I suspect many of us would find such a product or application useful from time to time. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 with 1000 extensions
On 7/3/13 6:50 am, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. I would caution against that approach. Analogue to Digital conversions often seem to have 'problems' - mostly related to hangup detection and/or echo. If you really do want to use analogue phones, then use a good quality channel bank to bring the analogue extensions into Asterisk, not low-end ATAs. You also have to consider the value of your time. There's little point shaving a few pounds (or dollars, or euros) from the hardware cost if it's going to double the configuration time. And using 'cheap' components will add to your ongoing support burden for the system. Cheap != good value for money. Personally, I'd consider using something like the Snom 710. They aren't the cheapest SIP phones by any means, but they do have a very good remote provisioning and configuration system, which will substantially reduce the work you need to do in configuring handsets. If your budget won't stretch to the Snom units, the Yealink range as suggested by another poster might be worth looking at. I believe their cheapest (is it the T18?) SIP endpoint can be had for around 35GBP - I don't know what pricing is like in your local currency of course. I believe Yealink do also have a fairly reasonable remote provisioning system, but unlike the Snom system, I can't claim to have used it in anger. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Exiting the queue doesn't work
On 4/3/13 12:27 pm, Gertjan Baarda wrote: After extensively googling the issue, I've found everything (also bug related), accept my answer. What am I missing here? It sounds like the call is being caught by a retry cycle on the queue. Try adding n to your queue command from your dialplan. Also worth making sure you have retry=0 in your queue config. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
On 18/2/13 5:39 pm, Administrator TOOTAI wrote: on incoming call we have exten = 100,n,Dial(SIP/Handset_102SIP/Handset_103SIP/Handset_104,,) and always only Handset_102 is ringing, we receive busy back from the 2 others but they are not. Any clue? It depends which base station you're using - some of the earlier ones only supported one or two simultaneous SIP calls (remember dialling counts as a call, even if it's not answered). I seem to recall the N300IP (the one we use) supports 3 concurrent SIP calls. The easiest workaround is probably to create a fourth SIP account called '102_103_104' or something that's set to ring all 3 handsets on the Gigaset web interface. You can then Dial(SIP/Handset_102_103_104) substitute the SIP account you created above from Asterisk instead. A cautionary note with Gigasets in general: they claim each base station will support up to 7 handsets. In my experience, things start to get a bit flaky above 3 or 4 handsets (specific handsets not ringing periodically, etc.), so I suspect the base station might be CPU limited at some point, especially if you're asking it to use an expensive (computationally) codec like G.729. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
On 17/2/13 5:02 pm, Administrator TOOTAI wrote: customer102/Handset_102 xxx.yyy.zzz.153 D N 5062 OK (80 ms) customer103/Handset_103 xxx.yyy.zzz.153 D N 5062 OK (70 ms) customer104/Handset_104 xxx.yyy.zzz.153 D N 5062 OK (66 ms) That's perfectly normal with these phones, and shouldn't pose a problem. As you see, all handsets are identified with the same port, which means that on incoming call to one handset or when transfering a call with the asterisk transfer feature, all 3 handsets are ringing :-( You can specify which SIP account correlates to each handset in the Gigaset web interface. Go to Settings - Telephony - Number Assignment You want Handset 1 to use Connection 'Handset_102' for outgoing calls and for incoming calls (untick everything else except this for incoming calls). Likewise Handset 2 should use Connection 'Handset_103' for outgoing and incoming (again, untick everything but this option). Rinse and repeat for other handsets. I can confirm it does work properly - we have dozens of clients with Gigaset phones and separate SIP registrations per handset. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] SayDigits
On 8/2/13 12:11 pm, Doug Lytle wrote: Is there a way to slow down or speed up the speed at which SayDigits So, I'd have to say no. I suppose potentially you could re-record the sound files to 'say' each digit faster (and with shorter rolloff at the end of each word), then put those into a separate [language] folder in /var/lib/asterisk/sounds, then use those instead in your dialplan. You might even be able to process the existing recordings using your favourite audio editing tool to speed the sound files and reduce the rolloff at the end. No guarantees it'll sound any good, mind. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Wierd question - Give me your opinion please
On 5/2/13 11:45 am, Jared Baxley wrote: The closest building is 950 ft, the second is 1850 ft. These two buildings are connected via LRE's using existing 6 Pair, Unfortunately re cabling isn't an option. Other buildings are even further from the office, about 15 or so scattered about that only require 1 phone each. Have you considered running your own SHDSL between the sites, i.e. run a small DSLAM in the main building? That'd give you IP connectivity in the remote buildings which could be used for both phones and also general net access if required. You'd then avoid having to worry about analogue phones at all. A friend did this down the length of a heritage railway as they already had cable running the length of their tracks, and I believe it was fairly successful. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect a POTS robo alert dialer to asterisk with email notification
On 3/2/13 4:59 pm, David Smiley wrote: I finally found the perfect solution for me:http://www.amazon.com/La-Crosse-D111-101-E1-WGB-Wireless-Monitor/dp/B0081UR76G/ref=dp_ob_title_def The device is $69, plus $10/month for alerts. And I get to monitor the temperature online, which is a great bonus. Working on the assumption that you already have internet access at the property in question, I do wonder whether you might be better off with something network-based rather than phone based. You should be able to pick up a network temperature sensor relatively inexpensively, which you could in turn use to fire HTTP requests to a server under your control (even if it's at your home). You could then store temperature stats in a database and set up your own triggers to do something as and when they drop below a certain threshold. In my $dayjob I've set up a number of similar systems at bird hides in various national parks/wetlands here in the UK for similar purposes (with the addition of the ability to pull off CCTV images). These are running programmable ICs which make a simple HTTP call to a webserver running SQL over a 3G data SIM every hour. I doubt it'd be difficult for you to knock up something similar for your property. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] RTP timeout if the asterisk box behind NAT
On 3/2/13 11:38 pm, bilal ghayyad wrote: What should I do? Given that you said: This problem was not appearing when Asterisk machine was having static real IP address because I was enabling the rtptimeout paramters. I do believe the solution is simple: put it back on a public IP. For what it's worth, we have dozens of clients with boxes on RFC1918 IPs and we don't see this issue, so I wonder if it's something 'special' your NAT router's doing to mess up RTP traffic. It's probably worth trying a different router (ideally different make/model) and see if that's any better. And it's always worth disabling any SIP ALG present in the router - they seem to do nothing but break things. (as a random aside, has anyone *ever* come across a scenario where a SIP ALG in a consumer router has actually helped?) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] RJ11 x RJ45
On 1/2/13 12:41 pm, Luis H. Forchesatto wrote: Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 está crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu não sei como deve ficar. Li alguns manuais na internet mas não entendi ao certo como tem que ser feito. Thanks to Google Translate, this apparently says something like: How do you make a RJ45 connector on one end and one RJ11. I intend connect to an ATA line on a plate Khomp KFXO IP. The tip having RJ45 connector is crimped with the sequence 568B and will be connected to the Khomp plate, but the tip RJ11 I do not know how it should be. I read some manuals on the internet but did not understand exactly how it must be made. Generally speaking the line pair are on pins 4+5, usually the blue pair. So if all you need is a single line pair, you should be able to just wire up the blue pair to the centre pins on your RJ11 connector. Alternatively, you can cheat, and just use an RJ11 - RJ11 cable - these usually fit just fine into an RJ45 socket. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] AGI command
On 20/1/13 4:15 pm, Eric Wieling wrote: Personally, I use the PHPAGI library and don't worry about all the low level stuff. This. It also gives you a nice logging function you can use to output debug information to the asterisk CLI so you don't have to kill and start asterisk interactively. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Voicemail and recordings storage: best practices
Greetings list, I'm currently building a new cluster to replace our ageing Asterisk 1.4 infrastructure - it's easier to start from scratch then migrate users across than it is to upgrade 1.4 to 1.8 in situ. Anyway, it got me thinking about audio recordings in a multi-server environment and whether there was a better way to do it. On our existing 1.4 cluster we NFS mount voicemail and recordings directories from another server (or more accurately a master/backup pair of servers) into each asterisk server. I'd say it's worked 'okay' - but since less than 10% of our users regularly use call recording, it's never really reached a point where I/O throughput has been an issue. So, since I have the opportunity to build up the new cluster from scrach, I thought it was an ideal opportunity to do a quick straw poll of the list and see what approaches others are using to store voicemail and recordings, and to make those available across a multi-server environment. Let the discussions begin. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] recrding calls
On 19/1/13 1:25 am, Joseph wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
How do I email these file?
You probably want to use MixMonitor() instead of Monitor():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Mixmonitor
One of its options allows you to execute a command at the end of
recording, which you can then use to call a script to handle your
recordings however you wish.
Kind regards,
Chris
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Re: [asterisk-users] DIDForSale spam
On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote: Unfortunately, there is a fine line between being a forum where people can exchange ideas, and being a forum where people can find asterisk consultants, and both don't seem to co-exist well together. Isn't this precisely the raison d'être for [asterisk-biz]? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Need help designing implementation
On 29/11/12 6:33 pm, Dyweni - Asterisk-Users wrote: I want to setup two Asterisk servers that are linked to each other: - The first server would be my external (public) server and would live in a real data center. The second server would be my internal (private) server and would live in my house. - The external server would receive all incoming calls and handle the voice mail stuff. - The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server. That all seems perfectly doable. - If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can't reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system. I don't think that's doable without quite a lot of work - but others may be able to advise further. To elaborate a little, it's easy to detect whether a route is usable when a call is placed, but detecting a call failure *during* the call is much more difficult. - If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call. That's easy, though remember asterisk does things in seconds rather than rings. You should also remember there's a delay in processing the call through the mobile networks before the phone actually starts ringing - in the UK that averages around 7 seconds between the call being sent to the mobile network from your server, and the phone ringing. - I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM). I would like specify in a white list specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents). Shouldn't be difficult. - I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages. I believe this is doable in the newer versions of asterisk, but not the older versions. Again, someone else will hopefully chip in here, since our stuff is still running 1.4 :-) Is all of this possible? If not, which part's are not (and how much work do you think would be needed to make those parts work)? As is so often the case, (almost) anything is possible if you're prepared to spend time doing it. How much is worth doing depends on your time, and what else you might prefer to be doing with it... FWIW, you might want to think about whether you actually need a separate asterisk box at home. In my experience, unless you have many dozens of extensions, you're almost better off (and certainly no worse off) connecting your SIP devices at home (assuming you're using SIP) directly back to the * server in the datacentre. One less box to maintain, and things like MWI will just work without having to play with the messaging interfaces. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Noise on phones while speaking...
On 13/11/12 6:52 pm, Carlos Chavez wrote: Why would a SIP to SIP call have this noise? Check to see what random stuff they have on their desk. We've regularly seen things like mobile phones (or cellphones to those of you across the pond :-) ) causing interference with VoIP phones. We've also of late seen some (especially Iiyama) monitors doing likewise - I suspect they have a fairly noisy 240v-12v transformer inside. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Sending calls from behind NAT
On 13/11/12 9:31 pm, Leighton Brennan wrote: It looks like you need to enable the sip application layer gateway or ALG on your router Quite often the reverse is true. Most routers (at least those I've used) seem to have such a lousy implementation of a SIP ALG it's often far better to just disable it and do your own NAT fixups in Asterisk (as others have indicated in previous posts). In fact, it's now the first thing we advise clients to do when they report call problems or one-way audio: disable the SIP ALG in your router. Sadly, there are also quite a few routers out there now that have ALGs that can't be disabled (or that make it extremely difficult to disable them). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 Support from Digium
On 4/11/12 8:37 pm, Danny Dias wrote: For example, if i install a FreePBX/Elastix I'd be very surprised (no, actually, I'd be *amazed*) if Digium were prepared to provide support on a product from a third party, which is what FreePBX and Elastix effectively are. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Multitenant opensouce application
On 31/10/12 6:20 pm, Darin Iv wrote: is another way to build Multi Tenant system, have to design like Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. snip Is there any particular reason why it needs to be _exactly_ like that? FWIW, we use companyA-201, companyB-201, companyA-202, companyB-202 as our SIP usernames. Each companyX then has its own extensions.conf file which contains a specific [companyX] context. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] high capacity analog - sip gateway
On 25/10/12 9:49 pm, Justin Killen wrote: What would be the advantage of using 100 single units vs. just buying VoIP phones? That doesn't seem very cost effective to me in the long run. In older buildings with existing single pair cabling, there might not be a great deal of choice. We were faced with a similar scenario at a hotel in Lincolnshire a few months ago - listed building, lots of old pre-ethernet cable, no likelihood of being able to replace the cable. In answer to the OP, I concur with suggestions for 24-port channel banks - you really don't want one or two devices responsible for all 100 extensions. I would not encourage individual SPA or PAP units - it'd be an administative (and cabling) nightmare - it's bad enough with a dozen of the things. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
On 2/10/12 6:51 pm, Carlos Alvarez wrote: Your traffic level, number of concurrent calls, etc would help us know what sort of carrier you should be talking to. Equally important, your geographic location, and the geographic location to which most of your calls are made will be useful in helping list members advise you. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Maximum messages in voicemail
Greetings list, I've seen a few errors recently in our logs along the lines of: [Sep 10 17:41:41] WARNING[6719] app_voicemail.c: Save failed. Not moving message: destination folder full. maxmsg in voicemail.conf is set to 1000. I've checked the mailboxes on the server in question, and the maximum number of messages in any account is 535. The filesystem in question is less than 50% full; likewise available inodes are plentiful. Does asterisk have a hardcoded maxmsg figure that supercedes voicemail.conf at a certain point? Is there any way of telling which mailbox the warning relates to? Anything else worth checking? Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Maximum messages in voicemail
On 10/9/12 6:48 pm, Danny Nicholas wrote: What flavor of asterisk? Realtime or just files? Post your voicemail.conf. Flat files, latest 1.4.x Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Japanese voicefiles
On 23/8/12 5:26 pm, Adrian Marsh wrote: I've a few questions around languages I'm on 1.4.18 (old yes I know, but upgradings not an option just yet). I've downloaded the gsm Japanese files from ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place I've found that when I switch to jp, and play some of my own voicefiles in Background and Playback, that it chooses the /var/lib/asterisk/sounds/jp folder files and plays them, but, voicemail doesn't seem to do this, instead it picks the English files (although the debug output says its using 'jp'). We use a similar method to play British English sounds - they're in /var/lib/asterisk/sounds/britishfemale - and voicemail seems to pick them up correctly. Have you made sure to specify language = jp in the relevant places? You need to do it in whichever module is originating the call to Voicemail - so if it's a SIP client, you'd probably do it in sip.conf, but if it's an incoming call, it's probably easier to do it in extensions.conf. FWIW, this is also using an old version - 1.4.21, so unless something's changed between .18 and .21, it should work with your setup. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] qualifysmoothing
Greetings list, I have a scenario where half a dozen phones at a site appear to be dropping offline for a few seconds every few hours, but the connection between them and the asterisk server remains up. It's been suggested to me that the problem might be to do with qualify - which is enabled in this case. However, I don't really want to disable it if at all possible - it's a very good early warning indicator of network problems, and has often proved useful in diagnosing network faults - especially with end users' *DSL connections not provided by us. AIUI from the documentation, qualifysmoothing effectively averages the last two qualify results. Is there any way to increase this, so a device won't be considered unavailable until, for example, 3 consecutive qualify packets have been missed? Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Digium IP Phone D40 quality, very bad
However, there's no reboot button in the web GUI of the phone. I have no experience with the phone in question and so will make no comment regarding the OP's original problem, but the absence of a software reboot function in the web GUI seems to be a pretty major oversight in my view. I do hope that one gets added to the we should really add this to the firmware ASAP list :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] What TTS to use?
On 26/7/12 11:08 am, Ishfaq Malik wrote: I'm thinking about deploying TTS onto our asterisk servers and was just wondering which ones people use and like... We've tried Festival, Cepstral and Ivona. Ivona was by far and away the best. If you need free (or very low cost) then your only real option is Festival. Just make sure to pick one of the newer voices. This was all based on UK English. If you're after something else, you may find different results. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] What TTS to use?
UK English is exactly what we're after. Did you try flite at all? No, I wasn't aware of flite when we ran these tests. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] click to call
On the subject of click to call - admittedly not necessarily what the OP was after - I had some marketing blurb from VMware about Zimbra 8 this morning. Apparently one of the new shiny features is integrated C2C (and other unified comms stuff). Has anyone had a chance to play with the SDK as yet? Would be quite fun to see if Asterisk could be integrated (visual voicemail and the like). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
On 10/7/12 7:46 pm, Tim Nelson wrote: Not to sound like a broken record or anything... but I'd say give Elastix a go. It is top notch in terms of release quality and features. And, being based on FreePBX, you can set it to 'Device and User' mode instead of the default extensions mode so users can 'hotdesk' between phones. We have a number of deployments using the FPBX 'Device and User' mode in a similar manner to that the OP requested, and they seem to work fairly well. On 10/7/12 7:58 pm, Patrick Lists wrote: Or you could just install a bare Asterisk and slap FreePBX on top of it Personally, I'd go down this route, especially if you're already familiar with Linux (which I'm guessing you are if you're used to working with bare Asterisk). This way you can choose the distro you like best rather than having to adapt to whatever the all-in-one maintainer has chosen to use. It also opens up options if you find you need to run other packages on the same server at any point. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] BT Fibre and 2701HGV
On 29/6/12 9:59 am, Ishfaq Malik wrote: Does anyone have any experience of connecting SIP phones to an asterisk server through the 2701HGV router that BT supply with their Infinity product? Good luck with that. The BT 'Home Hub' and 'Business Hub' routers they supply with retail ADSL and FTTC products seem to have a very poorly written SIP ALG in them that cannot be disabled [0] easily. This seems to play havoc with any attempt to hook up SIP devices - even just one handset. [0] I have read that it's possible to disable the ALG through a telnet session to the router, but it's somewhat fiddly and doesn't always 'stick' - so has to be repeated whenever the router is restarted. In my experience it's far easier just to replace the router with something competent. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Intro to DECT vs IP
On 29/6/12 11:16 pm, Michelle Dupuis wrote: Can you really mix match any base station with any DECT handset? Yes and no. Do handsets have proprietary features which only work with their own basestations? (eg: transfer between handsets)? Yes. And that's the 'no' part of my answer above - whilst they may make take calls, you might well lose additional functionality. Transfer hasn't been a particular problem (in my experience, it's better to use the native asterisk functions for this on DECT phones), but call lists most definitely have been an issue. Can i buy a good base station and get cheap Costco Dect handsets? As above, if you weren't worried about all the features, quite probably. But reasonable Gigaset DECT handsets designed for the base aren't exactly expensive - I think the C610H is around the 30GBP mark - substantially less if you're ordering quantity. And I've seen older models for substantially less - I picked up a batch of new - but old model S450s for around 30GBP for 6. I don't think I've seen DECT units in Costco for much less than 20 GBP. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Intro to DECT vs IP
On 30/6/12 12:12 am, Michelle Dupuis wrote: I like the look of the C610H. Is there a matching DECT base station by Gigaset? I use the N300IP. Supports 3 active SIP calls I believe - and yes, does have multiple SIP accounts (6, if I recall correctly). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Running Asterisk on VMware ESX
On 8/6/12 9:17 pm, Hiers, Richard wrote: I don't expect to need to use any special hardware, just a sip trunk over our broadband connection. We have about 150 phones at present. Is ESX a viable platform for us? And second, what is the recommended virtual configuration (mem, cpu, etc.)? Any other considerations? I think the concern expressed about running Asterisk on a virtualised platform is more to do with the impact the other load on the host machine might have on your Asterisk VM. If you're using ESX in a shared hosting environment where you have very little control over the other workload on that host, then sooner or later there's a risk your VM is going to experience spikes in latency. On the other hand, if you're running a virtualised platform internally where you can control precisely the load on the host machine, then you'll probably find you're fine. FWIW, we run Asterisk under Xen in production. Some of the VMs have well over a thousand connected SIP devices and we've yet to encounter significant problems. But we're able to control the other VMs on the hosts very precisely: the only other VMs running on those hosts provide low-load services such as rsync for remote backup (which is only used late at night when call load is low on the Asterisk VMs). Running Asterisk in a VM, even if it's the only VM on that host, does give you some considerable benefits in the event of host machine failover: hardware independence and live migration are the two that spring immediately to mind. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] OT - Incoming fax cuts ADSL line
I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used I too have seen this, and also with credit card processing machines in shops that 'dial' the merchant bank to process transactions (in effect a modem). It can sometimes (but not always) be resolved by running two microfilters in series on the 'voice' side of the line, i.e. line - microfilter - microfilter - fax. I've also (here in the UK) seen it resolved through the use of higher quality faceplate splitters rather than the often low-cost units supplied free with consumer ADSL modem/routers. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfer not working
On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Reliable SIP Trunk Provider
On 15/3/12 3:45 pm, Jake Wicke wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. You should probably let the list know what region/country you're in, as you'll want to be as close (i.e. low latency) to your trunk provider as possible. shameless plugIf you're in the UK, we (Minotaur IT) are a SIP trunk provider, and I'd like to think we support Asterisk and offer decent support :-) /shameless plug Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Low cost BRI gateway
Greetings list, I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately, everything I've seen seems to want to do lots of other things - registering handsets, IVRs, voicemail, etc. I only want it to present an ISDN BRI as a SIP account - I have an asterisk server for the other stuff. :-) In any other environment I'd just use one of the Digium ISDN PCIe cards, but in this case the ISDN lines come into one building and the asterisk servers are in the other building across the road, and there's no copper link between them. Any suggestions gratefully received. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] AGI: blocking script until playback complete
Greetings list, I've done AGI scripting before, but in the past I've always wanted control to be returned to the dialplan as soon as possible. However, today I have a scenario where I want the script to remain running during the playback of a file so that I can read DTMF at the end of playback. However, doing this: GET DATA en_welcome 5000 6 Results (correctly) in the following in the asterisk console: -- SIP/a.b.c.d-dc027b50 Playing 'en_welcome' (language 'en') But the AGI continues to run on after this point, not waiting for either the sound file to be played, nor for the expected 6 DTMF digits. Adding a simple 10 second sleep/wait to the AGI allows the sound file to be successfully played back. I'm sure I must be missing something very obvious, buy my google-fu is failing me this afternoon. Suggestions gratefully received :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] AGI: blocking script until playback complete
On 22/2/12 2:55 pm, Danny Nicholas wrote: You don't state the Asterisk version you are running, but personal experience tells me you'd better invest in some Rogaine if you're depending on the built-in stuff from AGI for DTMF input. I have personally wasted weeks trying it. Sorry, should have said, latest 1.4 release. Care to elaborate a little on the issues you found when you tried it? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] AGI: blocking script until playback complete
On 22/2/12 2:50 pm, Zohair Raza wrote: Try passing escape character GET DATA $filename $timeout $max_digits $escape_character Not sure I follow - according to the docs, there is no parameter $escape_character The problem seems to be that GET DATA returns control to the script before the audio file has even played, let alone any DTMF tones have been entered. I would have expected script execution to be blocked until the result from GET DATA was available. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] AGI: blocking script until playback complete
On 22/2/12 3:39 pm, Ron Bergin wrote: Have you tried increasing the timeout value in the command? To me, it appears to be too short. Try setting it to 5. Thanks for the suggestion - I tried 5, but no difference, it's not waiting at _all_ for the playback to complete. Just for giggles, I tried exactly the same test on a 1.8 box I have for testing, and the same problem occurs. I'm sure I must be doing something wrong here :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 1.8 minimum modules/configuration
Greetings list, Has anyone compiled (or could point me at) a list of the minimum required modules and conf files for a very basic 1.8 deployment? We have lots of 1.4 boxes in production, and I'm currently setting up a pair of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the existing 1.4 boxes. All the new installs need to do is receive calls via IAX and send them out via SIP to the 1.4 boxes. No ISDN or analogue channels, no voicemail, no conferences, codec translations, etc. - just the minimum number of modules necessary for basic IAX to SIP routing. Suggestions gratefully appreciated, otherwise I guess I'll try disabling everything, then gradually enabling modules as needed :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Preserving CDR(accountcode) in Local channels
Greetings list, Whilst running through a routine check of some CDRs, I've noticed that the originating channel's accountcode isn't preserved on creating a local channel. For example, if we start with: exten = 123,1,Set(CDR(accountcode)=foo) exten = 123,n,Queue(bar,nrtw,,,) And the queue 'bar' is defined as follows: [bar] member = Local/4...@outbound member = Local/5...@outbound member = Local/5...@outbound The 'accountcode' field is empty in the CDRs, even if I add /n to the queue members. So, a couple of quick questions if I may: 1) Is this behaviour expected? 2) How would one go about the above scenario whilst preserving the 'accountcode' field? Thanks in advance! Regards, Chris -- _ -- 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] Dahdi 2.3.0.1 fails to compile in Xen DomU
Greetings list, I've compiled and installed dahdi countless times on standalone machines, but recently I've been trying to compile Dahdi in a Xen DomU without much success. The errors I'm seeing are as follows: /var/tmp/portage/net-misc/dahdi-2.3.0.1/work/dahdi-linux-2.3.0.1/drivers/dahdi/zaphfc/base.c:1689: error: 'modes' undeclared (first use in this function) /var/tmp/portage/net-misc/dahdi-2.3.0.1/work/dahdi-linux-2.3.0.1/drivers/dahdi/zaphfc/base.c:1689: error: (Each undeclared identifier is reported only once /var/tmp/portage/net-misc/dahdi-2.3.0.1/work/dahdi-linux-2.3.0.1/drivers/dahdi/zaphfc/base.c:1689: error: for each function it appears in.) /var/tmp/portage/net-misc/dahdi-2.3.0.1/work/dahdi-linux-2.3.0.1/drivers/dahdi/zaphfc/base.c:1689: error: 'modes' undeclared here (not in a function) etc. The DomU in question is running Gentoo Linux with a 2.6.34 Xenified kernel. I've tried stepping back to 2.6.32 and 2.6.31 kernels, as well as dropping dahdi back to 2.3.0 and 2.2.0, all without success. I wouldn't be in the slightest bit surprised if I'm missing a kernel option somewhere, but I'm not sure where to start (or how best to diagnose such things). Is there a list of required kernel options for dahdi published anywhere I could consult, or has anyone else come across similar errors before? Any suggestions gratefully appreciated. Thanks in advance. Regards, Chris -- _ -- 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] OT: Bandwidth calculations
ISP 10% rule is what you are asking about expected that average usage is 10% of total subscribers with bursts higher But remember to plan well for those bursts and ensure you have sufficient excess capacity. Certain events can have a significant effect on your burst pattern: some fellows are kicking a ball around in South Africa for three weeks, which is having an understandable effect on bandwidth usage globally. Same happened a couple of years ago during the Olympics. Regards, Chris -- _ -- 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] Blind transfer feature
Am running 1.4.18 at the moment, and am trying to implement inline blind transfer. I have : [featuremap] blindxfer = *6 ; Blind transfer Do remember that asterisk needs to be in the media stream for this to work, so you'll want to make sure (in the case of SIP devices) you've set canreinvite=no You might also want to increase the feature code timeout (both activation and interdigit) - I think the default is something like 500ms, which most users find far too short to use reliably. Regards, Chris -- _ -- 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] Small PC to build and run Asterisk
I'm looking to build an Asterisk box that can run at a remote location. We've used the Asus eeeBox (desktop version of their little netbooks) quite successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD. Generally we run Gentoo Linux with Asterisk 1.4.latest, but no reason why you couldn't run another version you're more comfortable with. For 3 concurrent calls, even a machine of this spec might be overkill, but it's a good general-purpose server to have on-site at a remote location and might be useful for other things (general fileserver, rsync backup server, etc.). The eeeBox also has the advantage of being cheap (quite probably cheaper than smaller/lower power units), which means keeping a spare around in case of hardware failure isn't an unrealistic option. Regards, Chris -- _ -- 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] Small PC to build and run Asterisk
Actually, the Atom seems to be surprisingly powerful. We have a couple of Atom boxes with transcoding and conferences enabled without issue. I wouldn't pretend it'll cope with hundreds of conference participants, but with ~10 or so it seems to be fine. Likewise with transcoding - we've only really tested up to ~30 channels with G.711 to GSM, not any of the heavier CPU workload translations (e.g. iLBC or G.729). For a small to medium office (e.g. 30 extensions, 10 concurrent calls) it works fine, even with a little conferencing and transcoding. Regards, Chris -- _ -- 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] [OT] Asterisk 1.6 and DECT Phones
looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. We've been using the Siemens Gigaset range for a few years now (specifically C475IP and S685IP). Not had any major problems with them. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons -- _ -- 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] Snom Phones Registration/Failover Feature
I was reading in the documentation about the SNOM phones (mainly 300) but I did not find anything in the users-pdf's or on there knowledgebase/website which would tell me if this is possible, there is something for failover configuration but it is not explained at all. It's highly appreciated if someone with insight could explain to me or point me to the right documentation on how/if this work Looking at the docs on the Snom Wiki, it would seem that if you set up the primary server as Server 1, the secondary as Server 2, and set the Failover Identity option on ID1 to ID2, then it should work as you describe. I must confess I've never actually tried failover directly from the Snom phone (instead using Heartbeat/LVS as you suggest), but I see no reason why it shouldn't just work. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 870
Anybody tried one with Asterisk yet ? Views ? Apparently not available until the end of August. We've certainly used the Snom 820 with Asterisk without any issue in the past, and since both are based on (largely) the same software, I doubt there'll be any major problems with asterisk interoperability. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM Phones Displays NR Frequently
First things first. You are running /very/ old versions of firmware - particularly on the 300 and 320. Upgrade them. I've been running 7.3.14 for some time without a problem, though it appears that 7.3.23 is now out. I concur about upgrading the software, but I'd stick with 7.3.14 for now - at least until there's more feedback on stability of .23 and .24. On the OP's original question, I have noticed Snoms (of any version) aren't very good at handling DNS failure - they tend to cache DNS lookup failure almost indefinitely. If DNS lookups to your registrar occasionally fail, you might want to specify registrar via IP rather than by name. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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
