Re: [asterisk-users] Asterisk 16 Certified 16.8 and MagicJack Incoming Calls
It seems that if there is a pause in the auto attendant longer than a second this problem occurs. I have this for an extension in my extensions.conf file: exten => 2799,1,GotoIf($["${CALLERID(num)}" = "${EXTEN}"]?500) exten => 2799,2,Dial(PJSIP/${EXTEN},14,tr) exten => 2799,3,Dial(PJSIP/${EXTEN},1,tr) exten => 2799,4,BackGround(abandon-all-hope) exten => 2799,5,BackGround(dial-here-often) exten => 2799,6,Wait(2) exten => 2799,7,BackGround(gambling-drunk) exten => 2799,8,BackGround(you-seem-impatient) exten => 2799,9,BackGround(nobody-but-chickens) exten => 2799,10,BackGround(tt-somethingwrong) exten => 2799,11,BackGround(tt-weasels) exten => 2799,12,Voicemail(${EXTEN},ug(15)) exten => 2799,13,Voicemail(${EXTEN},bg(15)) exten => 2799,14,Hangup exten => 2799,500,VoicemailMain(${CALLERID(num)}) If I change the Wait to 1 the MagicJack will hear everything. If I change it to 2, nothing is heard from that point on. Chris On 6/1/20 8:43 AM, Chris Dos wrote: > I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and > converted form SIP to PJSIP using the python script as a start and then > mofiying from there. I ran into an issue when testing that incoming calls > from MagicJack would go silent after about 10 seconds. This happened while > in the automated attendant area. This problem did not occur with Asterisk > 13 LTS. I reverted PJSIP back to SIP and the problem still occurred, so > that was not it. > > We connect to Flowroute for our SIP provider. I added "force_avp = yes" to > the Flowroute endpoint section in the pjsip.conf and the problem appeared to > be solved after I tested it a dozen times. However, this morning this issue > has reappeared. Any thoughts on what might be causing this? > > My Flowroute pjsip.conf config: > [transport-udp] > type = transport > protocol = udp > bind = 0.0.0.0 > tos = cs3 > > [reg_us-west-wa.sip.flowroute.com] > type = registration > retry_interval = 20 > expiration = 120 > transport = transport-udp > outbound_auth = auth_reg_us-west-wa.sip.flowroute.com > client_uri = sip:12345...@us-west-wa.sip.flowroute.com > server_uri = sip:us-west-wa.sip.flowroute.com > > [auth_reg_us-west-wa.sip.flowroute.com] > type = auth > password = XXZZXXZZXXZZ > username = 12345678 > > [reg_us-west-or.sip.flowroute.com] > type = registration > retry_interval = 20 > expiration = 120 > transport = transport-udp > outbound_auth = auth_reg_us-west-or.sip.flowroute.com > client_uri = sip:12345...@us-west-or.sip.flowroute.com > server_uri = sip:us-west-or.sip.flowroute.com > > [auth_reg_us-west-or.sip.flowroute.com] > type = auth > password = XXZZXXZZXXZZ > username = 12345678 > > [reg_us-east-nj.sip.flowroute.com] > type = registration > retry_interval = 20 > expiration = 120 > transport = transport-udp > outbound_auth = auth_reg_us-east-nj.sip.flowroute.com > client_uri = sip:12345...@us-east-nj.sip.flowroute.com > server_uri = sip:us-east-nj.sip.flowroute.com > > [auth_reg_us-east-nj.sip.flowroute.com] > type = auth > password = XXZZXXZZXXZZ > username = 12345678 > > [reg_us-east-va.sip.flowroute.com] > type = registration > retry_interval = 20 > expiration = 120 > transport = transport-udp > outbound_auth = auth_reg_us-east-va.sip.flowroute.com > client_uri = sip:12345...@us-east-va.sip.flowroute.com > server_uri = sip:us-east-va.sip.flowroute.com > > [auth_reg_us-east-va.sip.flowroute.com] > type = auth > password = XXZZXXZZXXZZ > username = 12345678 > > [flowroute] > type = aor > contact = sip:12345...@us-west-wa.sip.flowroute.com > > [flowroute] > type = identify > endpoint = flowroute > match = 147.75.60.160/28, 34.210.91.112/28, 34.226.36.32/28, 147.75.65.192/28 > > [flowroute] > type = auth > username = 12345678 > password = XXZZXXZZXXZZ > > [flowroute] > type = endpoint > context = from-trunk > dtmf_mode = rfc4733 > allow = !all,ulaw > direct_media = no > from_domain = us-west-wa.sip.flowroute.com > tos_audio = ef > tos_video = af41 > ; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with > no audio after a few seconds. > force_avp = yes > auth = flowroute > outbound_auth = flowroute > aors = flowroute > t38_udptl = yes > t38_udptl_ec = fec > > [anonymous] > type=endpoint > context = anonymous > allow = !all,ulaw > -- _ -- 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
[asterisk-users] Asterisk 16 Certified 16.8 and MagicJack Incoming Calls
I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there. I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds. This happened while in the automated attendant area. This problem did not occur with Asterisk 13 LTS. I reverted PJSIP back to SIP and the problem still occurred, so that was not it. We connect to Flowroute for our SIP provider. I added "force_avp = yes" to the Flowroute endpoint section in the pjsip.conf and the problem appeared to be solved after I tested it a dozen times. However, this morning this issue has reappeared. Any thoughts on what might be causing this? My Flowroute pjsip.conf config: [transport-udp] type = transport protocol = udp bind = 0.0.0.0 tos = cs3 [reg_us-west-wa.sip.flowroute.com] type = registration retry_interval = 20 expiration = 120 transport = transport-udp outbound_auth = auth_reg_us-west-wa.sip.flowroute.com client_uri = sip:12345...@us-west-wa.sip.flowroute.com server_uri = sip:us-west-wa.sip.flowroute.com [auth_reg_us-west-wa.sip.flowroute.com] type = auth password = XXZZXXZZXXZZ username = 12345678 [reg_us-west-or.sip.flowroute.com] type = registration retry_interval = 20 expiration = 120 transport = transport-udp outbound_auth = auth_reg_us-west-or.sip.flowroute.com client_uri = sip:12345...@us-west-or.sip.flowroute.com server_uri = sip:us-west-or.sip.flowroute.com [auth_reg_us-west-or.sip.flowroute.com] type = auth password = XXZZXXZZXXZZ username = 12345678 [reg_us-east-nj.sip.flowroute.com] type = registration retry_interval = 20 expiration = 120 transport = transport-udp outbound_auth = auth_reg_us-east-nj.sip.flowroute.com client_uri = sip:12345...@us-east-nj.sip.flowroute.com server_uri = sip:us-east-nj.sip.flowroute.com [auth_reg_us-east-nj.sip.flowroute.com] type = auth password = XXZZXXZZXXZZ username = 12345678 [reg_us-east-va.sip.flowroute.com] type = registration retry_interval = 20 expiration = 120 transport = transport-udp outbound_auth = auth_reg_us-east-va.sip.flowroute.com client_uri = sip:12345...@us-east-va.sip.flowroute.com server_uri = sip:us-east-va.sip.flowroute.com [auth_reg_us-east-va.sip.flowroute.com] type = auth password = XXZZXXZZXXZZ username = 12345678 [flowroute] type = aor contact = sip:12345...@us-west-wa.sip.flowroute.com [flowroute] type = identify endpoint = flowroute match = 147.75.60.160/28, 34.210.91.112/28, 34.226.36.32/28, 147.75.65.192/28 [flowroute] type = auth username = 12345678 password = XXZZXXZZXXZZ [flowroute] type = endpoint context = from-trunk dtmf_mode = rfc4733 allow = !all,ulaw direct_media = no from_domain = us-west-wa.sip.flowroute.com tos_audio = ef tos_video = af41 ; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with no audio after a few seconds. force_avp = yes auth = flowroute outbound_auth = flowroute aors = flowroute t38_udptl = yes t38_udptl_ec = fec [anonymous] type=endpoint context = anonymous allow = !all,ulaw -- _ -- 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] Cisco 3950 ip phone
Hi, https://community.cisco.com/t5/ip-telephony-and-phones/cp-3905-asterisk/td-p/1995981 The phone does work, you do need to TFTP the configuration files to the phone though. Doesn't look like custom firmware is required. -- Chris. On Fri, Apr 12, 2019 at 3:29 PM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Friday 12 April 2019 at 15:24:27, Gokan Atmaca wrote: > > > > Please give us a link to a datasheet for that device. > > > > Hello > > > > > https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/u > > nified-sip-phone-3905/data_sheet_c78-651588.html > > Ah - that explains why I couldn't find a model 3950 phone :) > > Well, it uses SIP, so it should work as well as any Cisco phone does on > Asterisk. > > > Antony. > > > On Fri, Apr 12, 2019 at 3:58 PM Antony Stone wrote: > > > On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote: > > > > Hello > > > > > > > > Can I use Cisco 3950 on Asterisk ? > > > > > > Please give us a link to a datasheet for that device. > > -- > BASIC is to computer languages what Roman numerals are to arithmetic. > >Please reply to the > list; > please *don't* CC > me. > > -- > _ > -- 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 -- Regards, Chris Knipe -- _ -- 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] 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] 1000 analogue lines with asterisk
+1 spending money to get that many fxs ports is going to negate any savings of reusing analog phones instead of buying ip phones 1000 analog ports sounds like hell and if it was me I would be embarrassed to have a setup like that tied to my name if I was a consultant etc. Someone will come in after you and ask who set it up and the customer will say you :) On Feb 17, 2016 4:23 AM, "A J Stiles"wrote: > On Wednesday 17 Feb 2016, Goke Aruna wrote: > > Hello all, > > Can someone recommend what hardware to use for a 1000 analogue line > > capacity asterisk PABX? > > > > Regards > > A PCI express card with four primary rate ISDN ports, each linked up to a > channel bank, will give you 120 analogue lines. So you will need nine such > cards; and for reasons of simple numbers of slots on a motherboard, they > will > have to be split among three or more servers, linked to a gigabit switch. > > You might end up getting a better deal if you bought 1000 hardware SIP > phones. > (You also would probably increase your personal indispensability factor, > into > the bargain .) > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Panic Button SMS Asterisk Integration
Has anyone done any integration of USB, etc. panic buttons and Asterisk? The basic idea would be to have a USB based panic button[1] along with a bit of code which would trigger a group SMS or perhaps a pre-recorded call to a group. Kind regards, Chris [1]http://www.amazon.com/StealthSwitch-Pro-USB-Foot-Pedal/dp/B00MI6K77K -- _ -- 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 and Vitelity's vMobile service
I'm trying to configure my Asterisk machine to work with Vitelity's vMobile service. I can place calls to the vMobile device and it rings as expected. However, I have no audio in either direction. There's no NAT involved though. My asterisk machine has a public IP address with port 5060 and the full range of RTP ports open. (It does have a second NIC and some SIP clients on the private side but I'm not using those for my testing.) Here's the relevant parts of my SIP. Anyone see any obvious problems? Not much documentation around on vMobile. I'd love to see a working config if someone has one. Thanks! [general] nat=no externaddr=my_public_ip localnet=192.168.1.0/255.255.255.0 register = username:sec...@inbound23.vitelity.net:5060 bindaddr = 0.0.0.0 srvlookup=yes alwaysauthreject=yes transport=udp directmedia=no allowguest=no ; Vitelity vmobile SIP user [my_did_here] type=friend dtmfmode=rfc2833 host=my_did_here.mobilet100.sipclient.org context=vmobile-out defaultuser=my_did_here fromuser=my_did_here trustrpid=yes sendrpid=no secret=super_secret_password disallow=all allow=ulaw,g726,gsm -- 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] 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] Google Voice
I'm using chan_motif with Asterisk 11. It still works. I actually received an email from google yesterday that there had been no traffic on my number lately so the number would be reclaimed. I had switched my outgoing away from GV several months ago when they were supposed to discontinue the service. I switched back to it yesterday and have made several calls. No problems. On Sat, Jan 17, 2015 at 7:35 AM, CDR vene...@gmail.com wrote: Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -- _ -- 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 -- 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] Smartphone Mobility App?
Scott, Somehow never noticed this setting before, I have tested it and it really works great for the forwarding to cell situation we described. Thanks alot for pointing me in the right direction. chris On Fri, Dec 19, 2014 at 9:39 AM, Scott Griepentrog sgriepent...@digium.com wrote: The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their cell phone or the business DID. The easiest way to solve that is to have an audio prompt announce the calls that were passed through from the business DID before connecting the call through. That does require using a follow-me approach instead of forwarding, but is easily done by just changing the confimration prompt. On Fri, Dec 19, 2014 at 8:29 AM, chris tknch...@gmail.com wrote: Anyone found any good smartphone apps that connect with their asterisk boxes that provides basic mobility features? The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their cell phone or the business DID. We also would like to give user ability to control DND and forwarding of their extension from the smartphone. I know there are many cloud service providers with a offering like this but we are not looking to change our service infrastructure but rather looking for just a software product that connects to our existing asterisk systems and provides this functionality. We would ideally like something for both iphone and android but the immediate need is for iPhone Curious to hear what people have tried, their experiences, etc. We are open to both free/open source as well as commercial software as long as it is multitenant or scalable beyond single server. TIA, 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Smartphone Mobility App?
Anyone found any good smartphone apps that connect with their asterisk boxes that provides basic mobility features? The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their cell phone or the business DID. We also would like to give user ability to control DND and forwarding of their extension from the smartphone. I know there are many cloud service providers with a offering like this but we are not looking to change our service infrastructure but rather looking for just a software product that connects to our existing asterisk systems and provides this functionality. We would ideally like something for both iphone and android but the immediate need is for iPhone Curious to hear what people have tried, their experiences, etc. We are open to both free/open source as well as commercial software as long as it is multitenant or scalable beyond single server. TIA, 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] 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 -- 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] 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
[asterisk-users] Call Flow Documentation Tools
Hello, I have been researching software for documenting pbx call flow paths and I was just wondering if anyone out there is using anything they have found particularly useful or cool. I am looking for something preferably visual that the average end user can follow. So far the best thing I have come up with is making a diagram with a decision tree in visio but its very time consuming to build this by hand for every customer. We would like to be able to provide every customer a diagram so they can easily understand the path that a call takes, what conditions are checked and what actions are taken based on those conditions. A large portion of my asterisk installs are for non profit or charitable organizations so while I'm not completely fixed on a free solution, if it isn't free the cost needs to be relatively low or at least be a multi-tenant solution that could at least be used for multiple customers. For most of our installs we manage everything via CLI, but for a few orgs with tech savvy people we have been able to setup freepbx for them and let them make simple changes. I was thinking with freepbx maybe there could even be a module that takes the freepbx configuration and generates visuals based on reading the configuration, this would be really slick although not a complete answer as we have many installs that do not have freepx. Anyway, just wanted to get some input from others and put my ideas so far out there, if you have any recommendations or experiences to share feel free to reply on or off-list. Thanks! 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] 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
[asterisk-users] AUTO: Chris Douglas is out of the office (returning 09/16/2013)
I am out of the office until 09/16/2013. I will be out of the office and will have minimal access to email and voicemail. If you need immediate assistance, please contact the Pioneer I.S. Help Desk at 316-688-8777, 800-613-9382, or via intercompany dialing using the internal directory at http://directory.pioneer.world. There you can leave an emergency message for the on-call technician. Otherwise I will respond as soon as I can. Thanks, Chris Douglas Technical Services Manager Pioneer Balloon Company tel. 316-688-8648 fax. 316-691-6901 Note: This is an automated response to your message asterisk-users Digest, Vol 110, Issue 14 sent on 9/13/2013 12:00:01 PM. This is the only notification you will receive while this person is away. -- _ -- 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 -- 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] Am I being hacked?
On Mon, Aug 19, 2013 at 2:40 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 08/19/2013 08:10 PM, Eric Wieling wrote: One of Asterisk's dirty little secrets is that it does not show the source IP when a device or hacker tries sending a call without registering. The rejection message in the logs do not show the IP of the attacker. Yes it sucks, yes it has been that way for many many years. Are you aware of a patch that would show the source IP in the console and logs? I do something like this: 1. turn up the logging 2. add foo like this in my dial plan: exten = _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN}) exten = _.,n,Log(NOTICE,Anonymous peer IP: ${CHANNEL(peerip)}) exten = _.,n,Set(DID=${IF($[${EXTEN:1:2}=]?s:${EXTEN})}) exten = _.,n,Goto(s,1) 3. do some bar like this in my fail2ban filter: VERBOSE.*SIP/HOST-.*Received incoming SIP connection from unknown peer VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice' (language '.*') NOTICE.* .*: Anonymous peer IP: HOST NOTICE.* .*: Failed to authenticate device .*\s?\sip:.*@HOST\.* and that handles most of the hacking attempts I see on my system. I think it may be possible for the second line to catch some false matches, but I have not seen any issues with our system thus far. Kind Regards, Chris PS. Feel free to comment on what is wrong with this and be sure to include the right way to do it. :-) -- _ -- 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] 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] FTP server down?
Not sure if this is the right place to mention it, but . The server downloads.asterisk.org was refusing FTP connections last night, and still does not seem to be accepting them this morning. FTP may not be modern or trendy, but the ability to navigate around folders textually is nonetheless extremely handy when using a machine with no X server, from its own console. -- AJS Thanks for your concern and suggestions. Unfortunately, the decision was made to only offer our public downloads (e.g. downloads.digium.com, downloads.asterisk.org) over the HTTP protocol, not the FTP protocol, quite some time ago. As stated in Kevin Fleming's announcement on July 26th, 2007 to the Asterisk Announce mailing list, this was done primarily for reasons related to our marketing department ( http://lists.digium.com/pipermail/asterisk-announce/2007-July/85.html). If your response was misunderstood, please let us know and provide clarification. Thanks again! -- Chris Hozian Digium, Inc. | Network and Computer Systems Administrator 445 Jan Davis Drive NW - Huntsville, AL 35806 - US main: +1 256 428 6000 | fax: +1 256 864 0464 -- _ -- 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] Adjusting confbridge call quality
ulaw On Wed, Jul 10, 2013 at 7:40 AM, basteon bast...@gmail.com wrote: Hi, What codec do you use with yours subscribers? On 9 July 2013 23:45, Chris Gentle gent...@gmail.com wrote: Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? -- 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 -- _ -- 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 -- 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] Adjusting confbridge call quality
On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth mr...@imminc.com wrote: The sampling frequency for u-law is 8,000 Hz. You can't produce a recording with higher quality than the source, so you'd have to switch to a wideband codec to improve the conferences and recordings [1] [2]. OK, thanks for the info. I'm perfectly OK with 8,000 Hz except that I'm feeding the audio into a conference room from a microphone. chan_alsa actually is the first client to connect to the confbridge and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just wondered if there way some way to improve that. -- _ -- 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] Adjusting confbridge call quality
OK, thanks for the advice. No, there's no filter so I'll look into that. On Wed, Jul 10, 2013 at 3:02 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 07/10/2013 06:46 PM, Chris Gentle wrote: [snip] and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just wondered if there way some way to improve that. The distorted S and F are prevented by a pop filter in front of the mic. Are you using a pop filter? Also if you are using a cheap mic, do yourself a favor and invest in a decent mic. It will make a world of difference. Regards, Patrick -- _ -- 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 -- 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] Adjusting confbridge call quality
Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? -- 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] 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] blocking spammer by callerID name
Google the number and you can probably find other complaints and possibly who it is. Not that it will matter, there's nothing you can do but block it. My approach to call filtering is: Deny All Allow Some I have a whitelist of callers I always want to accept that may include businesses outside my local area code. If my dialplan doesn't recognize the incoming number I send them to a voicemail mail where they have to press 5 to leave a message. That knocks out the robo dialers. Then I google the number and if it's a spammer, I add them to a blacklist where the call is dropped immediately. Really no point in playing funny or cute messages to them or even telling them they are blacklisted because it's usually an auto-dialer and a real person doesn't hear it anyway. On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote: I have a subroutine to block spammer by CALLERID(number) exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten = 4,n,Set(goaway=${CALLERID(number):0:2}) exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1) exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1) but I just got another spammer (automated calls) who rotates his callerID number that starts with valid area code so blocking by prefix is not practical but it seems to me he uses the same (or few same) caller name like: Brit. Columbia 16047726633 KHAN SHARON 16042984429 Brit. Columbia 16042231781 So I was thinking the same subroutine can be used to block by CALLERID(name), isn't it: exten = 4,n,Set(goaway2=${CALLERID(name):0:11}) exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1) exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1) The spammer is soliciting lowering credit card interest charges etc. anybody know who it is :-/ -- Joseph -- _ -- 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 -- 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] blocking spammer by callerID name
Yeah, probably wouldn't work too well in a business environment where you actually NEED to answer calls. I go to a lot of trouble to make sure people can't get in touch with me. :) I keep my blacklist and whitelist in AstDB. However, I maintain it in a bash script so that I can update the script and then rebuild the AstDB very quickly. If I lose my AstDB I can just rebuild it with the script. ; Check the Asterisk database for blacklisted number exten = s,n,GotoIf(${DB_EXISTS(blacklisted/${CALLERID(num)})}?blacklisted,s,1) Whitelist can be done the same way: ; Check the Asterisk database for whitelisted number exten = s,n,GotoIf(${DB_EXISTS(whitelisted/${CALLERID(num)})}?voicemail,abc,1) I have a [screened] context that screens the calls and prompts for pressing 5 [screened] ;{{{ exten = s,1,Zapateller() exten = s,n,Set(TSTAMP=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten = s,n,NoOp(${TSTAMP}) exten = s,n,Monitor(wav,${TSTAMP}-${CALLERID(num)}-screened,m) exten = s,n,Set(COUNT=1) exten = s,n(loop),WaitExten(1) exten = s,n,Background(privacy-screening-unidentified-calls) exten = s,n,WaitExten(.5) exten = s,n,Background(press-5) exten = s,n,Background(T-to-leave-msg) exten = s,n,WaitExten(3) exten = s,n,Set(COUNT=$[${COUNT} + 1]) exten = s,n,GotoIf($[${COUNT} = 3]?loop) exten = s,n,PlayBack(goodbye) exten = s,n,StopMonitor() exten = s,n,Hangup() exten = 5,1,NoOp(Pressed 5) exten = 5,n,PlayBack(tcg/pls-lv-msg-w-nam-phnnum) exten = 5,n,StopMonitor() exten = 5,n,GoSub(voicemail,tcg,1) exten = 5,n,Hangup() exten = i,1,Playback(option-is-invalid) exten = i,n,Goto(99,msg) ;}}} On Thu, Jun 13, 2013 at 2:55 PM, Joseph syscon...@gmail.com wrote: Thank you for input. Good idea, I like your approach with press number to leave a message, this will definitely cut the robo-calls voice-mail. Do you use database for white-list? Can you post a section of your dial plan that deals with blocking? This is a medical clinic so white-list, black-list is not a good solution but it might be good for home use. Thanks, -- Joseph On 06/13/13 14:30, Chris Gentle wrote: Google the number and you can probably find other complaints and possibly who it is. Not that it will matter, there's nothing you can do but block it. My approach to call filtering is: Deny All Allow Some I have a whitelist of callers I always want to accept that may include businesses outside my local area code. If my dialplan doesn't recognize the incoming number I send them to a voicemail mail where they have to press 5 to leave a message. That knocks out the robo dialers. Then I google the number and if it's a spammer, I add them to a blacklist where the call is dropped immediately. Really no point in playing funny or cute messages to them or even telling them they are blacklisted because it's usually an auto-dialer and a real person doesn't hear it anyway. On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote: I have a subroutine to block spammer by CALLERID(number) exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten = 4,n,Set(goaway=${CALLERID(number):0:2}) exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1) exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1) but I just got another spammer (automated calls) who rotates his callerID number that starts with valid area code so blocking by prefix is not practical but it seems to me he uses the same (or few same) caller name like: Brit. Columbia 16047726633 KHAN SHARON 16042984429 Brit. Columbia 16042231781 So I was thinking the same subroutine can be used to block by CALLERID(name), isn't it: exten = 4,n,Set(goaway2=${CALLERID(name):0:11}) exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1) exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1) The spammer is soliciting lowering credit card interest charges etc. anybody know who it is :-/ -- Joseph -- _ -- 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 -- 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] 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
[asterisk-users] Confbridge doesn't kick chan_local
I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin@conferences I join the ices user to the confbridge with a call file: Channel: Local/1000@conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: conferences Extension: 1000_ices Priority: 1 This all seems to work great until I need to kill the conference. My confbridge user profile sets all users up with end_marked=yes so that they should be kicked when the leader exits. The Local channel doesn't seem to exit properly: Hangup on console -- Stopped music on hold on Local/1000@conferences-;2 -- Local/1000@conferences-;2 Playing 'custom/thank-you.slin' (language 'en') -- Executing [1000@conferences:2] Hangup(Local/1000@conferences-;2, ) in new stack == Spawn extension (conferences, 1000, 2) exited non-zero on 'Local/1000@conferences-;2' == Spawn extension (conferences, 1000_ices, 4) exited non-zero on 'Local/1000@conferences-;1' [Jun 3 07:44:27] NOTICE[18237]: pbx_spool.c:402 attempt_thread: Call completed to Local/1000@conferences The bridge gets left in an odd state. Rather than the bridge being destroyed, a confbridge list shows the conference is still active with 0 users. I have to restart asterisk to clear it. I wonder if someone might take a look at my dialplan snippet below and see if I'm doing something wrong before I file a bug report. I'm using Asterisk 11.4.0. [conferences] ; {{{ ; this is where normal callers enter the conference exten = 1000,1,ConfBridge(${EXTEN},testfone_bridge,testfone_user,testfone_user_menu) exten = 1000,n,Hangup() ; this is where the conf leader enters exten = 1000_admin,1,Answer() exten = 1000_admin,n,Set(CALLERID(name)=ConfLeader) exten = 1000_admin,n,Set(CALLERID(num)=001000) exten = 1000_admin,n,Set(CONFBRIDGE(user,admin)=yes) exten = 1000_admin,n,Set(CONFBRIDGE(user,marked)=yes) exten = 1000_admin,n,Set(CONFBRIDGE(bridge,record_conference)=yes) exten = 1000_admin,n,ConfBridge(1000,testfone_bridge,testfone_user,testfone_user_menu) exten = 1000_admin,n,Hangup() ; this is the Local channel that connects to app_ices exten = 1000_ices,1,Answer() exten = 1000_ices,n,Ices(/home/asterisk/asterisk-ices-1000.xml) ;exten = 1000_ices,n,Hangup() ;}}} -- 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] Confbridge doesn't kick chan_local
On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan mjor...@digium.com wrote: (1) Verify that with all 'normal' channel drivers, such as chan_sip, that the Conference tears down correctly. OK, looks like this is the problem. Taking chan_local out of the picture, I tested it with an incoming SIP call and an incoming IAX call and sure enough, the tear down fails. So if I rely on end_marked=yes to kick all my conference participants when the leader exits, it fails like this: Hangup on console -- Bridge/0x2364be4-input Playing 'confbridge-leave.slin' (language 'en') -- Stopped music on hold on SIP/gent_2880-0002 -- SIP/gent_2880-0002 Playing 'custom/thank-you.ulaw' (language 'en') -- Executing [1000@conferences:2] Hangup(SIP/gent_2880-0002, ) in new stack == Spawn extension (conferences, 1000, 2) exited non-zero on 'SIP/gent_2880-0002' confbridge list shows this: Conference Bridge Name Users Marked Locked? == == 1000 0 0 unlocked Now my confbridge is in a bad state. I tested it on both 11.4.0 and 11.3.0 on two different boxes with the same results. If both (1) and (2) are successful, than there's some impact that the Ices application is having on the Local channel that is messing up the reference counting inside the ConfBridge. Otherwise, it's an error in ConfBridge. So what do you think? Should I file a bug? -- 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] 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
[asterisk-users] Help me understand these log messages
OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand that my iptables would have stopped this but I'm just trying to understand more about the problem. What other settings might have stopped this? Fail2ban was running but there were no failed registration type messages that would have triggered it. [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '972595595767' rejected because extension not found in context 'default'. [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '00972595595767' rejected because extension not found in context 'default'. [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '000972595595767' rejected because extension not found in context 'default'. [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '011972595595767' rejected because extension not found in context 'default'. snip -- 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] Help me understand these log messages
OK, I understand now. I didn't realize allowguest was on by default. I guess I should read more closely. Thanks! On Fri, May 31, 2013 at 5:15 PM, Yves A. yves...@gmx.de wrote: ... an anonyous (not registerted) sip user from 188.161.238.232 was trying to initiate a call to 9725955 and so on... you could enable sip tracing to get more information. maybe you should change the 'allowguest' option in sip.conf..? regards, yves Am 31.05.2013 23:57, schrieb Chris Gentle: OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand that my iptables would have stopped this but I'm just trying to understand more about the problem. What other settings might have stopped this? Fail2ban was running but there were no failed registration type messages that would have triggered it. [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '972595595767' rejected because extension not found in context 'default'. [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '00972595595767' rejected because extension not found in context 'default'. [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '000972595595767' rejected because extension not found in context 'default'. [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '011972595595767' rejected because extension not found in context 'default'. snip -- 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 -- _ -- 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 -- 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] 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] chan_alsa and confbridge
Answering my own question. Setting the following in alsa.conf fixed my problem: input_device=plughw:0,0 output_device=null Changing the input device to plughw helped some but didn't completely clear the audio up. Setting the output device to null did the trick. I'm wondering if there was some kind of interrupt hammering going on here with my particular hardware. Even before the audio completely fell apart I could hear some little pops that sounded like the interrupts were not being serviced fast enough. On Mon, May 6, 2013 at 8:31 PM, Chris Gentle gent...@gmail.com wrote: OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions. I am trying to configure confbridge to create a conference room with an audio stream coming from my sound card. The idea is for a group of people to be able to call in and listen to someone giving a speech but not necessarily interact. I've got confbridge configured and it seems to work when I connect via other SIP phones. To get the alsa input into the conference I configured the alsa module and did this at the console: console dial 100@conferences This seems to work, once I got my alsamixer stuff set right. However, within about 10 seconds the audio goes bad. Lots of distortion, echo, etc. So I recorded a snippet right out of the sound card and loaded it into audacity. The snippet was fine. No distortion at all. So the problem seems to be something in asterisk. Any ideas what I'm missing here? Is there a better way to do this? -- Chris -- 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] chan_alsa and confbridge
OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions. I am trying to configure confbridge to create a conference room with an audio stream coming from my sound card. The idea is for a group of people to be able to call in and listen to someone giving a speech but not necessarily interact. I've got confbridge configured and it seems to work when I connect via other SIP phones. To get the alsa input into the conference I configured the alsa module and did this at the console: console dial 100@conferences This seems to work, once I got my alsamixer stuff set right. However, within about 10 seconds the audio goes bad. Lots of distortion, echo, etc. So I recorded a snippet right out of the sound card and loaded it into audacity. The snippet was fine. No distortion at all. So the problem seems to be something in asterisk. Any ideas what I'm missing here? Is there a better way to do this? -- 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] 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
[asterisk-users] E911 Voip Trunking
During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind 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] E911 Voip Trunking
Section 6.5.2 (v4 interface) of NENA's v2 Interim Voip Architecture Standard shows a ladder diagram of their SIP flow which seems to match standard SIP. Maybe I'm oversimplifying it? [1] http://c.ymcdn.com/sites/www.nena.org/resource/collection/2851C951-69FF-40F0-A6B8-36A714CB085D/NENA_08-001-v2_Interim_VoIP_Architecture_i2.pdf On Fri, Apr 19, 2013 at 2:51 PM, Terry Brummell te...@brummell.net wrote: E911 does not follow the standard SIP RFC. That would be a good reason that they couldn't/wouldn't do it. Now that I say that I should qualify it and say NG911 (or ESINet) does not follow SIP RFC http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying your county is not using standard SIP for E911, it just wouldn't be considered NG911. -- *From:* Chris Nighswonger *Sent:* Fri 4/19/2013 11:41 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] E911 Voip Trunking During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if anyone else has attempted this. Kind 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 -- _ -- 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] E911 Voip Trunking
On Fri, Apr 19, 2013 at 8:59 PM, Nathan Anderson nath...@fsr.com wrote: On Friday, April 19, 2013 5:35 PM, Warren Selby wrote: There are E911 providers that offer this functionality. I know off the top of my head, 911Enable offers a service like this. A former client of mine that provided hosted PBX services had a contract with them. I'm sure there are other providers out there as well. Indeed. 911ETC is who we use, and is another example. Even if you could peer directly with your county's PSAP, in the case of 911, I think it is a way better idea to go with one of these specialty SIP-based E911 providers, for the simple reason that even if you only sell VoIP service to people residing within your county Actually we are not reselling service and the majority of our phones are stationary. The few mobile soft phones we run would not need 911 service since they also carry cell phones, the soft phones being mainly remote extensions. So it sounds like it is at least worth pursuing. Kind 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] 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] xmpp priority setting and GoogleVoice
On Sat, Mar 23, 2013 at 10:45 AM, Harley Peters har...@thepetersclan.com wrote: I had it set to 1 originally and it worked fine at first then suddenly stopped. It drove me crazy until I ran across this link: http://iprouteth0.blogspot.com/2013/01/new-thoughts-troubleshooting-google.html Set it to 127 and it has worked ever since. Yep, that's the same article I found and mentioned in original post. Thanks for posting the link. While the article was written for Asterisk 1.8 and Jabber, the same setting works in the xmpp.conf file. Very useful stuff. -- 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] xmpp priority setting and GoogleVoice
I just wanted to send out some information that will hopefully help others. I don't know, maybe I'm the only one that's been having problems with this. I've been pulling my hair out for a while wondering why Google would not send my incoming calls to my Asterisk box. The calls would just roll to voice mail and no packets ever reached Asterisk. This has happened on two separate Asterisk boxes and three different GoogleVoice numbers. I've been all through the GV web page settings but nothing I did changed anything. I figured it had to be something simple, and I was right. I finally ran across an article that talked about the priority setting in xmpp.conf. This is set to 1 by default, which is apparently the lowest priority setting. Apparently, GV routes calls to whatever session has the highest priority. If I understand correctly, being logged into gmail/gtalk has a higher priority, somewhere around 20. So Google calls were probably being routed to my logged-in session but I never saw them because I don't use Gtalk. I changed this value to 100, as suggested in the article, and incoming calls immediately started working for all three GoogleVoice numbers. The Asterisk 11 xmpp.conf sample file currently has this to say about the priority setting: ;priority=1 ; Resource priority I'm going to go out on a limb here and say that probably could use a bit more. This seems like a pretty important setting. Just my two cents ... Hope this helps somebody else. -- 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] Asterisk does not persist callgroup and pickgroup configuration.
Freepbx is over writing the conf files ? On Mar 15, 2013 8:11 AM, Luis H. Forchesatto luisforchesa...@gmail.com wrote: Greetings. I'm running asterisk here (elastix) and I have a few extensions configured in it. I have here two different callgroup/pickgroup where the extensions are configured in, but it doesn't work when I try do pickup a call. Looking the config file (sip_additional.conf) I see they are not configured with callgroup/pickgroup, the fields are empty. Manually inserting callgroup/pickgroup on the extensions worked just fine but the next day the configuration just vanished and the extensions was not working. Has someone a clue of whats going on here? -- Att.* *** Luis H. Forchesatto -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 GoogleVoice/Motif
I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk has been up for a while (usually about a day), outgoing calls through GoogleVoice fail to complete. I hear it ringing on my end but the caller never hears the phone ring. A simple restart of Asterisk seems to clear it up for another day or so. Has anyone else noticed this? -- 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] 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 11 GoogleVoice/Motif
Awesome, thanks. I'll give it a try. On Mar 11, 2013 4:56 PM, Joshua Colp jc...@digium.com wrote: Chris Gentle wrote: I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk has been up for a while (usually about a day), outgoing calls through GoogleVoice fail to complete. I hear it ringing on my end but the caller never hears the phone ring. A simple restart of Asterisk seems to clear it up for another day or so. Has anyone else noticed this? This is fixed in Asterisk 11.3.0-rc1. The Google XMPP server has become prone to disconnecting as of late, which triggers a bug in older versions where chan_motif ignores XMPP traffic. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 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 -- 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] DIDForSale spam
+1 On Jan 9, 2013 8:59 PM, Don Kelly d...@donkelly.biz wrote: Jai, It should not be necessary for me to remove my email address from your list. It should not be on there to start with—we do not have, and have never had, a relationship that justified you sending me email. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Jai Rangi Sent: Wednesday, January 09, 2013 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DIDForSale spam Guys, Since I am attached to did for sale: My apology to every one who received the DIDForSale 2012 Achievement email and you hated it. As a asterisk user my question will be. If some xyz company send you a so called spam email, what made you think that you should spam the mailing lists. I am sure we all get lots if spam emails every day. If you really got some time and talent, why don't you write some good tips and tricks about asterisk. Long story short We have a link where you can unsubscribe your email for any further communication. http://www.didforsale.com/unsubscribe.php or Send me your email address I will personally take care of that and will remove your email. This will take less than 5 seconds. I am sure there will be lot of arguments on why you should that and all. I will refrain myself on any further unproductive communication. Happy new year to you all. On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote: +1 here. On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote: On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote: What were the senders IP(s)? Will have to look it up when I get home. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 I have gotten hit with this twice so far. in March and Today: Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com 3/8/12 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com 1/9/13 UGH, when I asked in March where he got my email he said: Hi Chris, We got your contact from the Internet. Let me know the good time to talk about this in detail. Thank you, -Rohit Dhaka Obviously by harvesting these lists. I received 2 myself. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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] DIDForSale spam
There is a big difference between publicly posting offering services to the list and harvesting all the email addresses and them contacting everyone privately On Thu, Jan 10, 2013 at 5:32 PM, C. Savinovich c.savinov...@itntelecom.com wrote: Isn't this precisely the raison d'être for [asterisk-biz]? Oh my goodness!, the asteriz-biz? nooo, they will kill you if you try to post anything offering your services!... that list ceased to provide any value and died a long time ago precisely because its members ran each other away from it. A while back, I wrote a nice click-to-call service and I dared put a post indicating that I was offering it for a fee, and in no time they called me spammer. There is really no incentive to reward someone else's achievements, unless you tell them that you are given them your code for free, then they want it (totally contradicting the meaning of the word business). Christian Savinovich VoIP Telephony Consultant 646-982-3572 Original Message Subject: Re: [asterisk-users] DIDForSale spam From: Chris Bagnall aster...@lists.minotaur.cc Date: Thu, January 10, 2013 5:17 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing music through VoIP handsets while on hook
I've seen this implemented on polycom phones where a secondary extension is on the phone that is setup to auto answer and they have something on the PBX side that is configured to call some or all of the secondary extensions On Jan 10, 2013 8:28 PM, Carlos Alvarez car...@televolve.com wrote: This is something I've seen with some key systems and PBXs. When the phones are on-hook, they can play music throughout the office instead of having an overhead speaker system do it. Never heard of it being done with VoIP, but figured I'd ask if anyone else has. I don't see any way to do this on any phones I've looked at. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing music through VoIP handsets while on hook
Lol yes it was all local on a gigE network even :) I also didnt say it was the most elegant solution but it seemed to work well with them they even had grouped it into extensions and I'm sure you could even write some logic to make sure the calls are local On Thu, Jan 10, 2013 at 9:31 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Jan 10, 2013 at 6:42 PM, Christopher Harrington ch...@acsdi.com wrote: Wow, that seems wildly bandwidth inefficient. Is it possible to do multicast VoIP? Depends on whether the phones are local to the server. Unless you're looking at hundreds of phones, a 100MB network running 80k to every phone wouldn't even notice. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDForSale spam
On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote: What were the senders IP(s)? Will have to look it up when I get home. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 I have gotten hit with this twice so far. in March and Today: Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com 3/8/12 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com 1/9/13 UGH, when I asked in March where he got my email he said: Hi Chris, We got your contact from the Internet. Let me know the good time to talk about this in detail. Thank you, -Rohit Dhaka -- _ -- 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 for Razberry Pi
On Wed, Jan 2, 2013 at 9:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? apt-get install asterisk Does anyone know of any asterisk 11 packages for the Pi? I ended up compiling it myself this weekend. Took a while. -- 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