| 00:08.08 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
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| 03:11.31 | *** part/#asterisk NeonLevel (~NeonLevel@189.165.36.66) |
| 03:20.49 | asr33 | [TK]D-Fender: fixed it, I finally understand GotoIF (and in the process had an extremely entertaining evening :-) ) |
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| 04:44.40 | Jentrep | I'm guessing this is a good place to meet Telecom Engineers? |
| 04:49.16 | *** part/#asterisk Jentrep (62c5d1ec@gateway/web/freenode/ip.98.197.209.236) |
| 05:09.06 | carrar | only if you hang around |
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| 06:56.51 | linocisco | hi all |
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| 07:08.28 | schmidts | good morning |
| 07:26.04 | davlefou | good day all. |
| 07:26.37 | davlefou | how can i do to create an include context in mysql? |
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| 09:07.10 | CommaCrazy | hi all |
| 09:08.45 | CommaCrazy | if anyone is buying PBX-es or telephony platforms DO NOT, i repeat DO NOT BUY Topex (Rohde & Schwarz) ANYTHING |
| 09:09.16 | CommaCrazy | do not buy it, it is one of the crappiest telephony platforms that I have ever seen. |
| 09:09.21 | lorsungcu | rofl |
| 09:09.32 | CommaCrazy | it is so wrong on so many levels |
| 09:10.04 | CommaCrazy | the support is crappy as well |
| 09:10.26 | CommaCrazy | and it is all based on billing and mysql.... |
| 09:10.36 | CommaCrazy | if you delete a user, it is not deleted |
| 09:11.32 | CommaCrazy | you need to manually log in trough ssh and delete the user from mysql for it to be truly gone |
| 09:11.49 | CommaCrazy | and that is after you have deleted the user from the crappy web interface they got |
| 09:11.58 | CommaCrazy | can't do anything cli |
| 09:13.41 | CommaCrazy | and for a simple function of forward a call to another phone you have to have 2 routes set up in Topex to allow you to forward a call to that phone |
| 09:14.18 | CommaCrazy | you have to add like 5 different routes to make a single hunt group (which does not even work properly) |
| 09:14.43 | CommaCrazy | I repeat DO NOT buy Topex anything if you want to save your sanity |
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| 09:45.17 | mirela666 | lol thx for info :) |
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| 09:53.57 | CpuID | hey ppls... anyone here had asterisk on a stock debian (wheezy) do silly things like hanging etc? |
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| 09:55.10 | schmidts | CpuID i dont think it depends on debian, maybe you have some deadlocks. which asterisk version do you use? |
| 09:57.24 | ghost75 | wheezy is not stable |
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| 09:58.33 | CpuID | sec let me confirm |
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| 09:58.44 | CpuID | its 1.8.13 |
| 09:58.51 | CpuID | yea deadlocks sounds likely... |
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| 10:02.17 | ghost75 | use sqeeze |
| 10:06.14 | CpuID | hehe |
| 10:07.00 | *** join/#asterisk rox (~rox@212.30.81.3) |
| 10:07.13 | rox | hello |
| 10:07.56 | rox | is there a way to get information about the number of current calls waiting in a queue in dialplan? |
| 10:08.33 | rox | is the queue data accessible in some sort of variable or something like that? |
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| 10:10.49 | rox | i would need to do something like this: |
| 10:10.49 | rox | exten _X.,1,Execif($[queueinfo(myqueue,callswaiting) < 10]? queue(myqueue)) |
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| 10:18.09 | schmidts | rox why not using group and grouptcount function? |
| 10:19.31 | rox | schmidts: i will look into that |
| 10:21.09 | rox | schmidts: do you mean something like, when i queue calls, i should also add them to an arbitrary group and then use group count? |
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| 10:28.52 | con3x | I'm getting random crashes when I try to transfer incoming calls from phones using chan_skinny. Anybody got any clues? I've found a bug report that suggests it might be a problem with the driver. |
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| 10:32.52 | brian98 | Hello asterisk gurus |
| 10:33.01 | brian98 | quick bit of help with a script please? |
| 10:33.29 | brian98 | I've put asterisk in as a PRI to PRI gateway at the moment for a customer with a view to moving to SIP soon |
| 10:33.38 | brian98 | the pbx sends 4 digit CLI |
| 10:33.47 | brian98 | they have 3 ranges of 100 ddi's |
| 10:34.03 | brian98 | based on the first 2 digits of the cli the pbx sends I am fxing up to make it the correct cli |
| 10:34.11 | brian98 | but I think my script isn't correct |
| 10:34.56 | brian98 | I have 3 checks in the from-pbx context in extensions.conf |
| 10:35.05 | brian98 | exten => _X.,n,ExecIf($["${CALLERID(num):0:2}"="11"]?Set(CALLERID(num)=1614${CALLERID(num)} )) |
| 10:35.13 | brian98 | exten => _X.,n,ExecIf($["${CALLERID(num):0:2}"="22"]?Set(CALLERID(num)=1614${CALLERID(num)} )) |
| 10:35.21 | brian98 | exten => _X.,n,ExecIf($["${CALLERID(num):0:2}"="33"]?Set(CALLERID(num)=1603${CALLERID(num)} )) |
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| 10:35.56 | brian98 | even if the callerid(num) is 33 it seems to match all 3 - what am I doing wrong ? |
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| 10:57.08 | linocisco | hi. where can I buy T-shirt with asterisk logo? |
| 10:57.15 | *** join/#asterisk davlefouAMD (~david@41.225.241.191) |
| 10:57.16 | linocisco | hi. where can I buy T-shirt with asterisk logo in Malaysia ? |
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| 11:01.35 | linocisco | hi. where can I buy T-shirt with asterisk logo in Malaysia ? |
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| 13:48.56 | pbxbrian | Hey Up - anyone awake? |
| 13:50.17 | leifmadsen | I'm sure millions of people are awake :) |
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| 13:57.22 | [TK]D-Fender | </meta> |
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| 14:01.13 | pbxbrian | Smart arse! |
| 14:01.14 | pbxbrian | ] |
| 14:01.15 | pbxbrian | :) |
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| 14:10.31 | ghost75 | when i am speaking on phone i hear some tone when another call gets in, is it possible to show who is calling without picking up the other call? |
| 14:11.58 | con3x | ghost75: you can check who is calling using the Management Interface, what phones are you using? |
| 14:12.34 | ghost75 | analog phones |
| 14:12.58 | [TK]D-Fender | ghost75, Call-waiting-callerid <- |
| 14:13.12 | [TK]D-Fender | ghost75, What is that phone plugged into? |
| 14:13.21 | ghost75 | pap2t |
| 14:13.52 | [TK]D-Fender | ghost75, Should support the feature. Your PHONE has to as well. |
| 14:14.06 | ghost75 | its a crap phone |
| 14:14.10 | [TK]D-Fender | ghost75, Many older (and cheap) phones don't. |
| 14:14.26 | [TK]D-Fender | CW-CID was a new(er) feature |
| 14:15.31 | ghost75 | to display with ami on website makes no sense i think |
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| 14:17.20 | ghost75 | some cheapish cisco voip desk phone would do or they are special? |
| 14:17.49 | [TK]D-Fender | ghost75, Any SIP phone with a display should be capable of it |
| 14:18.10 | ghost75 | i remember something cisco phones needs license |
| 14:19.21 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nmyxlqabrmouitbz) |
| 14:19.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:19.27 | ghost75 | hmm if i start softclient on pc maybe should also do it |
| 14:19.34 | con3x | ghost75: I have ciscos that display it using SLA |
| 14:19.49 | con3x | but thats a bugger to configure |
| 14:19.57 | ghost75 | service level agreement? |
| 14:20.43 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:20.56 | con3x | Shared Line Appearances |
| 14:21.05 | [TK]D-Fender | con3x, He just wants to see the CID on his 2nd incoming call. This has nothing to do with SLA. |
| 14:21.28 | [TK]D-Fender | con3x, You are on a very different page here..... |
| 14:22.54 | ghost75 | so when i fire up zoiper this works out of the box or i need to add something |
| 14:23.01 | con3x | Was just mentioning, thats just the way I have it configured =/ sorry if it was misleading |
| 14:23.07 | [TK]D-Fender | ghost75, Soft-client, AMI monitoring app, XMPP message, etc |
| 14:23.16 | [TK]D-Fender | con3x, It is... your phone still has to support it. |
| 14:23.28 | ghost75 | ok again new terms |
| 14:23.35 | [TK]D-Fender | con3x, and SLA has nothing to do with contributing to the functionality he requested. |
| 14:24.34 | [TK]D-Fender | ghost75, Zoiper is just a softphone. Out of the box (it doesn't come in one) it is unconfigured. You'll have to set it up... jsut like any SIP device. You'll also have to make another peer for it and call them both in your dialplan.\ |
| 14:24.51 | ghost75 | yes this is already done |
| 14:25.38 | ghost75 | just if i need to do something with that: Call-waiting-callerid |
| 14:27.18 | ghost75 | but when i speak on analog phone the extension of softphone should be anyway not be blocked |
| 14:28.08 | [TK]D-Fender | ghost75, CW-CID is an analog thing. To a SIP phone every call is just a call. |
| 14:28.16 | ghost75 | k |
| 14:28.50 | ghost75 | any other cheapish voip phones around, that are usable? |
| 14:32.08 | ghost75 | phone directory on display would be nice |
| 14:33.09 | con3x | Ciscos' can be quite cheap to get second hand, and Digiums phones are rather nice |
| 14:33.31 | ghost75 | yes cisco around 50 |
| 14:34.18 | ghost75 | the phonebook on display is done over ldap? |
| 14:35.04 | [TK]D-Fender | maybe you should go read the specs on the actual models you are looking at. They aren't all the same you know... |
| 14:36.00 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 14:36.03 | con3x | ghost75: Web request in my case =/ 7960s' |
| 14:37.42 | ghost75 | so cisco are default sccp, not sip |
| 14:38.51 | con3x | Yeah, and if you want Sip you load a firmware image over TFTP. |
| 14:39.14 | ghost75 | and you need cisco account to get fw :< |
| 14:39.15 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
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| 14:39.37 | con3x | Yep, thats Cisco for you. |
| 14:40.14 | Penguin | "Need" is such a relative term. |
| 14:45.47 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 14:57.17 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx |
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| 14:58.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:01.00 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:01.01 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:01.18 | Katty | pokes Qwell |
| 15:02.21 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 15:02.26 | Qwell | dies |
| 15:02.51 | schmidts | /etc/init.d/Qwell restart |
| 15:08.17 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:08.47 | Katty | prods Qwellery |
| 15:09.26 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
| 15:13.56 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 15:28.01 | ghost75 | (15:40:31) Penguin: "Need" is such a relative term. <- so there are unofficial sites offering fw updates? |
| 15:28.12 | Penguin | Many. |
| 15:28.31 | Penguin | As illegal as it might be, they are out there. |
| 15:28.52 | *** join/#asterisk serafie (~erin@nat/digium/x-dfhdfhhxyxtafjen) |
| 15:28.53 | Penguin | Are you looking for something specific? |
| 15:29.03 | ghost75 | i think 7960 are most common or? |
| 15:29.16 | Penguin | They're common. I don't know about most common, though. |
| 15:29.45 | ghost75 | what means G appendix on modell nr |
| 15:29.51 | Penguin | Global |
| 15:30.09 | Penguin | As in, the buttons have images instead of words. |
| 15:31.26 | ghost75 | can get a used one starting 30 bucks |
| 15:31.43 | Penguin | I prefer to have a Global phone and then add the English sticker to go around the buttons (if the sticker isn't already on it). |
| 15:31.45 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 15:32.13 | Penguin | The 7960 with SIP is pretty basic and lacking features, but it does work for making and receiving calls. |
| 15:33.43 | rrittgarn | Personally as far as cheap phones go, i like the SPA 509g, but then again that 'cheap' to me is $150ish. Polycom 321 is $80 ish new. Both have freely available firmware though without the need for an account. |
| 15:34.09 | [TK]D-Fender | 7940/60 are also Cisco PoE only. Make sure it comes with a power-brick/injector or your PoE switch supports it.... (NOT 802.3af) |
| 15:34.25 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 15:34.31 | Penguin | "PoE only" is a stretch. |
| 15:34.48 | Penguin | PoE capable would be better. |
| 15:34.59 | ghost75 | i think i need only phonedirectory on display and be able to transfer calls, see who is calling when busy |
| 15:35.58 | ghost75 | directory goes over ldap right? |
| 15:36.11 | Penguin | Not that I know of. |
| 15:36.13 | [TK]D-Fender | Penguin, They don't have a direct power cord. |
| 15:36.19 | Penguin | Yes they do. |
| 15:36.43 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 15:36.45 | ghost75 | has external power supply |
| 15:37.10 | [TK]D-Fender | Penguin, Ah, I think I'd missed that one.... |
| 15:37.19 | Penguin | You can plug the power supply right into the 48V jack on the back of the phone, or you can use PoE on the spare pairs. |
| 15:37.42 | *** join/#asterisk fritz09 (~Adium@port-11115.pppoe.wtnet.de) |
| 15:37.46 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:37.56 | [TK]D-Fender | Penguin, Indeed somehow I'd missed that way back... |
| 15:38.08 | ghost75 | how i get then cidnames to the display? |
| 15:38.19 | Penguin | Send calls to the phone. |
| 15:38.20 | [TK]D-Fender | <ghost75> i think i need only phonedirectory on display and be able to transfer calls, see who is calling when busy <- no. |
| 15:38.45 | [TK]D-Fender | ghost75, Why on earth do I need a directory to see the number of the person who's calling me ... that comes WITH the SIP call? |
| 15:38.46 | Penguin | You don't need a directory on the phone at all. |
| 15:39.09 | Penguin | Directory is nothing more than a phone book. It stores names and numbers. |
| 15:39.18 | [TK]D-Fender | ghost75, Why would I need a directory to transfer calls? I can do that with a hook-flash on an analog phone. |
| 15:39.22 | ghost75 | uhm i mean like a phone book where you can select a number and call it |
| 15:39.28 | [TK]D-Fender | ghost75, Lay off the doobie.... |
| 15:39.36 | Penguin | To see caller id information, just look at the screen when a call comes to the phone. |
| 15:40.01 | [TK]D-Fender | ghost75, If you WANT a directory fine, but that is in no way a requirement for either of those 2 other functions. |
| 15:40.19 | ghost75 | i just puzzled around wrong terms maybe :) |
| 15:40.47 | Penguin | A directory can be nice if you need to store a lot of names and numbers. Otherwise, just remember them. |
| 15:43.29 | [TK]D-Fender | On most phones the thought of scrolling through 50 entries 1 or two at a time on a phone is a massive productivity loss. Print a page. |
| 15:43.35 | ghost75 | i have phonebook also in sql, maybe present that over ldap? or how this works? |
| 15:43.45 | *** join/#asterisk gg608f (~Adium@187.207.51.115) |
| 15:44.05 | [TK]D-Fender | ghost75, it works however you set it up to work. |
| 15:44.22 | [TK]D-Fender | ghost75, Go learn LDAP. Go learn the other DB you have. Neither of these are Asterisk. |
| 15:44.40 | ghost75 | but its the way this is done right? |
| 15:45.10 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-xfkrnnetfszrsftf) |
| 15:45.27 | [TK]D-Fender | ghost75, No phone cares about SQL. |
| 15:45.39 | ghost75 | of course not |
| 15:45.42 | [TK]D-Fender | ghost75, As for what YOURS even supports... that is another matter |
| 15:45.55 | [TK]D-Fender | <ghost75> but its the way this is done right? <ghost75> of course not |
| 15:46.03 | [TK]D-Fender | ghost75, You are answering your own questions here.... |
| 15:46.04 | *** join/#asterisk lorsungcu (~anonymous@mbb5036d0.tmodns.net) |
| 15:46.08 | ghost75 | was just asking which protocols are involved when doing such things |
| 15:46.17 | [TK]D-Fender | ghost75, Depends on the phone. |
| 15:46.43 | [TK]D-Fender | ghost75, Some use an XML pull live. Some download a fixed file on boot. Some use LDAP. |
| 15:46.56 | [TK]D-Fender | ghost75, Go read the spec on your phone. |
| 15:47.10 | ghost75 | k |
| 15:47.45 | [TK]D-Fender | Some you can use a X/HTML browser on the phone. |
| 15:50.11 | *** join/#asterisk lorsungcu_ (~anonymous@69.199.191.69) |
| 15:51.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
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| 16:00.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:04.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 16:21.10 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 16:21.57 | igcewieling | Does anyone know how/where Asterisk sets the Contact: header? Our provider needs it to be sent in a slightly different format than Asterisk is sending by default. |
| 16:23.03 | Penguin | What is it sending now, and what do they want you to send? |
| 16:23.44 | igcewieling | Sending Contact: <sip:+2129182000@209.220.119.5:5060> Needs to be <sip:+12129182000@209.220.119.5:5060> |
| 16:24.12 | Penguin | Looks exactly the same to me. |
| 16:24.13 | [TK]D-Fender | igcewieling, Depends what comm is sending it |
| 16:24.22 | [TK]D-Fender | Penguin, 1 in front of the number |
| 16:24.36 | igcewieling | [TK]D-Fender: "comm"? |
| 16:24.37 | Penguin | s/Looks/The format looks/ |
| 16:24.40 | [TK]D-Fender | <PROTECTED> |
| 16:25.10 | igcewieling | I managed to get the From header in the correct format by setting the CALLERID(num) |
| 16:25.49 | Penguin | Change the register statement to reflect the required data. |
| 16:25.58 | [TK]D-Fender | igcewieling, Describe the 2 ends and what action is getting that header |
| 16:26.05 | *** part/#asterisk fritz09 (~Adium@port-11115.pppoe.wtnet.de) |
| 16:26.07 | igcewieling | Penguin: We are not registering |
| 16:26.31 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:27.03 | igcewieling | [TK]D-Fender: side A is Asterisk. Side B is Satan aka Verizon Business SIP. They require the TNs to be in e164 format including the + and the country code. |
| 16:27.48 | [TK]D-Fender | igcewieling, should be fromuser IIRC for an outbound call where they don't reg to you. |
| 16:28.30 | igcewieling | I assume I can set that in the dialplan since it needs to be dynamic |
| 16:28.41 | [TK]D-Fender | igcewieling, If they did then the contact would have been provided by them |
| 16:28.43 | igcewieling | we have over 5,000 TNs |
| 16:29.07 | [TK]D-Fender | igcewieling, Contact is who "you" are. That really isn't supposed to be "dynamic" |
| 16:29.25 | [TK]D-Fender | igcewieling, Sounds like you may need a proxy-rewrite for this |
| 16:29.48 | [TK]D-Fender | igcewieling, Or time to make a lot of peers.... |
| 16:30.12 | igcewieling | changing the Asterisk source code would be easier than installing a proxy just for this |
| 16:31.00 | Penguin | Why would you need to change who you are dynamically? |
| 16:32.04 | igcewieling | looks like the carrier is wrong and I am able to use CALLERID(num) to set it. |
| 16:32.14 | *** join/#asterisk lorsungcu (~anonymous@69.199.191.69) |
| 16:32.17 | igcewieling | Penguin: we are an ITSP, we have thousands of TNs |
| 16:32.51 | *** join/#asterisk lorsungcu (~anonymous@69.199.191.69) |
| 16:33.57 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:34.51 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 16:49.48 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:51.22 | carrar | igcewieling, Are you saying Verizon will not accept a contact of +1 |
| 16:52.06 | igcewieling | carrar: no, I'm saying they REQUIRE the contact to be +1. |
| 16:52.08 | Penguin | He indicated that they needed +1 before the rest of the number in NANP. |
| 16:52.15 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 16:52.22 | carrar | So just send all cals with +1 |
| 16:52.26 | carrar | in the invite |
| 16:52.32 | igcewieling | The issue is a non-issue, VZ looked at the WRONG CALL (they do this way too often_ and we are in fact sending the calls out with +1 now. |
| 16:53.05 | carrar | We also use Verizon |
| 16:53.08 | igcewieling | Penguin: Technically they want e164 which is "+" and the country code and the local number. |
| 16:53.12 | carrar | among a lot of others |
| 16:53.29 | igcewieling | carrar: do you have problems with calls randomly disconnecting in the middle? We do, VZ says it is because we are not sending with a 1+ |
| 16:53.34 | carrar | no |
| 16:53.56 | carrar | Our verizon service has been extremely solid |
| 16:54.11 | carrar | and you pay out the nose for that too :) |
| 16:54.14 | Penguin | That's bull. Calls wouldn't suddenly die because you didn't earlier send the right contact data. |
| 16:54.41 | igcewieling | Penguin: Trust me, it is easier to do what they say than argue with them about it. I do agree, however. |
| 16:55.08 | igcewieling | carrar: you must have their BIA product. We were one of their earlier customers and have the SIA product. |
| 16:55.11 | carrar | verything we send to verizon, all fields are all +1 |
| 16:55.39 | igcewieling | carrar: *nod* that is what I now have set. VZ said we still were not sending the 1+ but they were wrong. |
| 16:55.46 | igcewieling | ..e.r +1 that is |
| 16:55.59 | carrar | we've been using them for at least since 2008 |
| 16:57.17 | *** join/#asterisk dolcea-sgs3 (~androirc@51-210-17-190.fibertel.com.ar) |
| 16:57.24 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:57.39 | igcewieling | carrar: we are trying to avoid paying though the nose for upgrading from SIA to BIA. SIA is *cheap* since you don't get transit via VZ only SIP |
| 16:57.59 | igcewieling | carrar: my boss said we started using VZB around the same time. |
| 16:58.18 | carrar | They had the longest interop list of any carrier so far |
| 16:58.20 | carrar | heh |
| 16:59.03 | carrar | had to write a bunch of custom code just them |
| 16:59.03 | *** join/#asterisk dolcea-sgs3 (~androirc@51-210-17-190.fibertel.com.ar) |
| 16:59.42 | igcewieling | I was not here for the VZB interop, but Level 3's interop list was pretty long. |
| 17:00.05 | carrar | yeah their's isn't too bad |
| 17:00.55 | igcewieling | Level 3 annoys me because, unlike VZB, if they don't find a valid ANI/etc they will bill the call to a "default" TN and we have no way to bill it back to the customer |
| 17:01.04 | igcewieling | VZB simply rejects those calls. |
| 17:01.31 | igcewieling | VZB lets you use Diversion |
| 17:02.26 | carrar | Diversions are fine for remote termination |
| 17:03.03 | carrar | but for LD carriers doesn't it doesn't matter so much |
| 17:03.17 | igcewieling | We only use them on forwarded calls. |
| 17:04.47 | igcewieling | We prefer to pass the original callerid on forwarded calls, Diversion lets us do that |
| 17:04.53 | carrar | Are you using asterisk as your "SBC" to your carriers? |
| 17:06.15 | igcewieling | carrar: not at the current time, but that is the long term goal. MAYBE with Kamailio in front of them. |
| 17:06.44 | carrar | we use opensips |
| 17:06.50 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:07.20 | igcewieling | *nod* All calls are routed based on DB lookups so even a sip proxy in front of the Asterisk boxes would have heavy lifting to do in the "dialplan" |
| 17:07.41 | *** part/#asterisk dolcea-sgs3 (~androirc@51-210-17-190.fibertel.com.ar) |
| 17:08.10 | carrar | a proxy in front should cache your whole db of DID's in memory |
| 17:08.28 | carrar | as well as where to pass teh call |
| 17:09.23 | carrar | as well as all your LCR tables |
| 17:09.48 | igcewieling | any yet, all that is already handled by our existing system which is well known to us and very flexible. |
| 17:10.41 | igcewieling | call comes in, look up the route by DNIS, if not found, look it up by ANI, process ECF, CFU, default dial, CFNA, CFBL, and DTO sending the call to the right peer each time |
| 17:13.14 | igcewieling | If we find issues with scaling we might add a sip proxy or more Asterisk boxes or both |
| 17:13.45 | igcewieling | Now if we can get our VZ SIA upgraded to SIA we will be able to stop proxying all RTP through our existing SBC |
| 17:13.54 | igcewieling | SIA upgraded to BIA |
| 17:15.21 | carrar | sip proxy is gonna scale more with less resources |
| 17:16.46 | igcewieling | I know, that is why we looked into it originally. |
| 17:18.09 | carrar | and you get to learn a whole new call processing language :) |
| 17:19.11 | igcewieling | that is the problem. We don't want to learn a whole new call processing language. 8-| |
| 17:19.35 | carrar | You could hire someone to do it for you in a few days |
| 17:19.50 | igcewieling | The hard part isn't the language it is having to actually understand SIP instead of letting Asterisk do all the magic. |
| 17:20.01 | carrar | thats also true |
| 17:20.19 | igcewieling | carrar: a few days? To port a thousand lines of code from the asterisk boxes to opensips/kamailio |
| 17:20.29 | carrar | but reading and understanding the RFC should been done anyways |
| 17:20.54 | carrar | depending on how well you document what you do |
| 17:21.01 | carrar | so maybe a few weeks |
| 17:21.33 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 17:21.48 | igcewieling | I'm sure we will have a sip proxy eventually. |
| 17:21.49 | carrar | It's just passing calls and loggin for the most part |
| 17:22.09 | igcewieling | "just passing calls", huh? |
| 17:22.09 | carrar | heh "just" :) |
| 17:22.36 | saint_ | hi all |
| 17:23.39 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 17:23.40 | *** join/#asterisk gusto (~gusto@bband-dyn184.95-103-31.t-com.sk) |
| 17:23.53 | gusto | hi folks |
| 17:29.41 | *** join/#asterisk navaismo (~Administr@189.191.20.105) |
| 17:36.38 | *** join/#asterisk Xaviertoor (~Xaviertoo@186.195.41.114) |
| 17:39.25 | igcewieling | carrar: do you also have service with Level 3? |
| 17:39.48 | carrar | GBLX |
| 17:39.59 | carrar | We're suppose to be turning up a L3 circuit here soon |
| 17:40.27 | carrar | L3 bought GBLX |
| 17:40.36 | leifmadsen | neat |
| 17:40.53 | leifmadsen | turning up circuits with L3 does require some legwork since you have to go through a certification process |
| 17:40.55 | igcewieling | once VZ and L3 merge a black hole will open up and destroy everything. |
| 17:41.06 | carrar | probably won't have to do that |
| 17:41.10 | carrar | since we're a existing customer |
| 17:41.25 | leifmadsen | if you don't make changes to the circuit you should be ok |
| 17:41.32 | leifmadsen | we need to do it to change capacity of the circuit |
| 17:41.38 | carrar | haha |
| 17:41.41 | carrar | thats so lame |
| 17:41.42 | leifmadsen | I love it :D |
| 17:41.44 | leifmadsen | agreed |
| 17:41.57 | leifmadsen | luckily i just have to manage it, not actually do the tests :) |
| 17:42.18 | carrar | I don't mind, I have a room full of Fax's, phones and crap to quickly do the testing in a day |
| 17:42.40 | leifmadsen | I have things I'd rather be building than testing circuits :) |
| 17:42.48 | carrar | me too |
| 17:43.32 | carrar | it's good practice :) |
| 17:44.18 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 17:44.21 | carrar | we're not so big where I can assign it to someone else |
| 17:44.38 | leifmadsen | I hear that |
| 17:44.53 | carrar | and I rather kinda do it myself |
| 17:49.21 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 17:51.26 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:00.21 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:07.06 | ghost75 | which isdn cards you recommend for business? |
| 18:07.24 | navaismo | Digium |
| 18:07.40 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 18:07.42 | igcewieling | Sangoma |
| 18:09.12 | ghost75 | eicon? avm? |
| 18:11.27 | carrar | Adtran |
| 18:11.34 | carrar | AudioCodes |
| 18:11.38 | carrar | why use a card? |
| 18:11.40 | carrar | heh |
| 18:12.26 | carrar | Crisco |
| 18:12.26 | *** join/#asterisk kontinuity (~kontinuit@122.167.231.88) |
| 18:13.14 | ghost75 | why not use card? |
| 18:13.31 | carrar | less overhead on your machine |
| 18:13.44 | ghost75 | instead use gateway? |
| 18:13.51 | igcewieling | ghost75: no sane person will recommend any card other than Digium and Sangoma. Obviously you could also us a SIP/PSTN GW like Adtran |
| 18:13.55 | carrar | yes |
| 18:14.23 | carrar | but if your only using a few T1's cards are fine |
| 18:14.44 | carrar | What does the future hold!! |
| 18:14.49 | ghost75 | well only like 4B channels or so |
| 18:15.04 | carrar | card |
| 18:15.17 | carrar | I like Digium cards |
| 18:15.41 | ghost75 | are digium cards still rebranded noname cards? i read something on an australien site |
| 18:16.25 | carrar | I don't think so |
| 18:16.35 | carrar | They have their name on the circuit board |
| 18:16.51 | carrar | least the ones we have do |
| 18:18.02 | igcewieling | ghost75: that was ONLY the X100P and has not been the case in 10 years. |
| 18:19.12 | carrar | I still have a few of those around someplace |
| 18:19.27 | *** join/#asterisk gr0mit (~tim@2a01:9c41:ffff:30:d4c4:7f24:e81a:ca8d) |
| 18:19.56 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 18:20.08 | carrar | Back when Mark would log into your box to see why something wouldn't work |
| 18:20.11 | igcewieling | I did until Katrina |
| 18:20.14 | *** join/#asterisk frek818 (~Adium@124.6.136.142) |
| 18:20.31 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 18:30.17 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 18:30.17 | *** mode/#asterisk [+o ChanServ] by rajaniemi.freenode.net |
| 18:30.31 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 18:30.55 | _Corey_ | ghost75: Digium cards are definitely not "rebranded noname cards"... ;) |
| 18:32.20 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 18:37.12 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:37.34 | Katty | QUICK! give me some stocking stuffer ideas. |
| 18:38.30 | carrar | soap |
| 18:38.33 | tzanger | my first answer would be legs but that's not what you're asking about |
| 18:38.37 | carrar | M&M's |
| 18:38.40 | carrar | pencils |
| 18:38.43 | carrar | pens |
| 18:38.50 | carrar | back scratcher |
| 18:38.51 | _Corey_ | rebranded noname cards perhaps? |
| 18:38.57 | tzanger | yes, chocolates are always good, my mom always got clementines for that |
| 18:38.58 | carrar | gift certs |
| 18:39.05 | tzanger | usb keys are useful |
| 18:39.25 | carrar | floppy discs |
| 18:39.38 | Katty | teehee floppy discs. |
| 18:39.54 | Katty | gift certificate is a good idea. |
| 18:40.14 | carrar | small little toys |
| 18:40.20 | Katty | it's for an adult. |
| 18:40.26 | carrar | mind puzzles things |
| 18:40.47 | Katty | a sports car loving, vmware administrator...who likes console games. |
| 18:41.05 | carrar | Book about KVM |
| 18:41.57 | tzanger | heh |
| 18:42.28 | carrar | espresso beans |
| 18:42.41 | Katty | YES |
| 18:42.44 | Katty | that is a wonderful idea. |
| 18:43.03 | igcewieling | thinkgeek.com usually has something interesting |
| 18:43.13 | tzanger | you think they'll ship on time? |
| 18:43.23 | Katty | not without a horrendious ammount of shipping |
| 18:43.28 | Katty | i'm thinking of a best buy run |
| 18:43.33 | Katty | tho they are awful. |
| 18:43.39 | carrar | Bake something |
| 18:44.13 | carrar | Every year a relative makes awesome fresh baked bread |
| 18:44.31 | *** join/#asterisk Nugget (~nugget@rennsport.macnugget.org) |
| 18:45.16 | Katty | HAI NUGGET |
| 18:46.31 | Nugget | hi hi! |
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| 18:52.15 | gusto | hm |
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| 19:01.04 | Penguin | It's snowing. |
| 19:01.56 | Robotman321 | I want snow.. |
| 19:02.10 | Robotman321 | this bubble called Chicago doesnt allow for snow lately.. |
| 19:02.54 | Penguin | I'm 300 miles south of you. |
| 19:03.00 | Penguin | (roughly) |
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| 19:47.49 | joako | I´m setting up a new Asterisk installation with Polycom phones. I went into the config files and edited the hostname of the old system to the new hostname. I visit the phone´s HTTP GUI and I can see the new host... but somehow some of the phones still register to the old installation. How can this be!? |
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| 19:59.01 | Penguin | Verify the host name is correct in DNS and in the phone's settings. Restart the phones. |
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| 20:05.20 | joako | Penguin: Interesting. Actually physically rebooting the phones works. sip notify, reboot through the HTTP GUI unregisters and registers the phone, but not to the new server. |
| 20:05.43 | joako | Finally is Set(CALLERID(name)=Something still valid for Asterisk 1.8? It doesn´t have any effect. |
| 20:08.12 | leifmadsen | it works fine |
| 20:08.16 | leifmadsen | depends on your network though |
| 20:08.22 | leifmadsen | and you may need sendrpid=yes enabled too |
| 20:08.35 | leifmadsen | if your syntax is exact, then you're missing a ) |
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| 20:09.44 | joako | Well its the exact config that was working in Asterisk 1.4 |
| 20:09.59 | leifmadsen | shrugs |
| 20:10.01 | leifmadsen | looks fine to me |
| 20:10.31 | leifmadsen | check UPGRADE.txt to make sure you've nto missed something. Should just need sendrpid=yes in sip.conf and Set(CALLERID(name)=Isuck) |
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| 20:12.35 | joako | I already had sendrpid=yes and I tried to change it to sendrpid = pai and there´s no UPGRADE.txt in AsteriskNOW |
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| 20:18.32 | Penguin | CALLERID(name) works just fine, and does what it is supposed to do. |
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| 20:22.03 | danfromuk | Does anyone know a voiceover recording studio in London which a client could use to make some ivr recordings? They want to use their own artist, they just need the equipment. |
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| 20:55.59 | jeffspeff | what's the difference between admin monde and marked mode in meetme |
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| 21:05.04 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.1.0 (2012/12/10), 10.11.0 (2012/12/10), 1.8.19.0 (2012/12/10), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:19.12 | jeffspeff | i'm messing around with meetme, using it in realtime with mysql. when i set the user pin and admin pin, and specify option w, it doesn't mark the admin user as the actual marked user. how can i change that? if someone enters the admin pin instead of user pin, i want them to be the marked user. |
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| 22:13.15 | SuperNull | yo momma is a sip packet. |
| 22:13.32 | SuperNull | one with many VIAs .. OHHHHHh |
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| 22:16.20 | saint_ | re |
| 22:16.41 | saint_ | if curl-devel was not installed on my machine, and i installed it after asterisk, do i need to configure && make && make install again ? |
| 22:17.17 | saint_ | when I try to use ${CURL(http://...)} it says Function CURL not registered |
| 22:17.29 | navaismo | yes you need to recompile |
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| 22:31.47 | jameswf | ~ |
| 22:35.26 | leifmadsen | saint_: yes, and then 'module load func_curl.so' |
| 22:49.09 | navaismo | Is there a way to reboot a Digium phone to factory default from WEBGUI or a command?? |
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| 23:10.41 | *** join/#asterisk Primer (~daniel@www.ceregatti.org) |
| 23:11.44 | Primer | Hi, I created a digitmap on my polycom 501 that goes: 100x|...the default.... but when I type 1004 it goes back to a dial tone and the display reads "Enter more digits". If I type 1002, 1003, 1005,...1009, it works fine |
| 23:12.03 | Primer | There is no other digitmap pattern that matches 1004. Anyone know what this might be? |
| 23:12.24 | Primer | I realize it's a bit off-topic here, but I figured this might be the best place to ask |
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| 23:39.27 | tompaw | Guys, how do I detect stalled calls in confbridges? |
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