| 00:02.55 | *** join/#asterisk jfindley (~jfindley@104-181-196-33.lightspeed.hstntx.sbcglobal.net) |
| 00:03.05 | jfindley | Any PRI gurus around? |
| 00:05.52 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 00:11.47 | [TK]D-Fender | Maybe a sub-guru will have you answer |
| 00:11.53 | [TK]D-Fender | just ask your question. |
| 00:14.29 | jfindley | I'm trying to figure out how to link to asterisk boxes via PRI |
| 00:15.12 | jfindley | I've got PBX 1 as NET with master clock and PBX 2 as CPE with slave clock, but I'm guessing I'm missing something |
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| 00:40.24 | [TK]D-Fender | You're definitely missing something |
| 00:40.55 | jfindley | Is it possible with a dual span card to connect both ports with a crossover cable and be able to dial out one span and answer on the other? |
| 00:40.57 | [TK]D-Fender | You haven't described any actual failure or problem. You haven't shown us configs or status dumps. You haven't told us what hardware you're using. |
| 00:41.11 | [TK]D-Fender | We don't see any call attempt |
| 00:41.15 | WIMPy | yes |
| 00:41.30 | WIMPy | But you need the correct crossover cable. |
| 00:42.25 | [TK]D-Fender | steps out for a few while actually useful data is hopefully being collected |
| 00:42.37 | jfindley | would a straight through cable work as well? |
| 00:42.46 | WIMPy | No |
| 00:43.13 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 00:43.22 | WIMPy | At the bottom. |
| 00:45.16 | jfindley | interesting... |
| 00:46.55 | jfindley | The type of cables I've been using mostly are flat grey straight through cables with RJ-45 heads and 8 wires. Are only 4 of them being used when going, e.g., from an adtran to an A102? |
| 00:47.30 | WIMPy | Only Gigiabit ethernet uses all 8. |
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| 00:48.17 | jfindley | So what's the difference between the adtran DSX-1 port and a port on the A102 connected to another port on the A102? |
| 00:48.51 | jfindley | sorry, success is had when going from the adtran to the A102, but no joy when going from one port on the A102 to the other port. |
| 00:49.23 | WIMPy | Unless you have a card where you can jumper the pin-out, they are wired as TE. |
| 00:49.52 | jfindley | So I'd also need a crossover cable for going from the A102 to another A102 in a different system? |
| 00:50.17 | WIMPy | I don't know what you can change there. |
| 00:51.23 | jfindley | I don't mind using a crossover cable, if that's what you mean by 'change there'. |
| 00:51.45 | *** join/#asterisk AviiNL (~AviiNL@h178213.upc-h.chello.nl) |
| 00:52.27 | WIMPy | Jike having jumpers or multiple ports wired differently. |
| 00:52.46 | WIMPy | But you easily see if the cable is correct on the LEDs. |
| 00:55.58 | jfindley | I'm going to make a crossover cable and see how that works. Thanks WIMPy! |
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| 01:10.13 | jfindley | hah! it works! |
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| 01:28.29 | snadge | hmm.. asterisk console is broken.. instead of scrolling up and down |
| 01:28.45 | snadge | it just rotates through the last commands |
| 01:28.56 | snadge | page up and page down seems broken.. wtf :p |
| 01:29.24 | snadge | restarted the terminal.. weird |
| 01:30.22 | josefig | hi, if i want to create a pattern to dial my voip provider should something like exten=>_Z.,1,Dial(SIP/provider/${EXTEN}:2,120,r) ? |
| 01:31.55 | [TK]D-Fender | no. there is an error in what you think is a proper variable reference |
| 01:32.04 | [TK]D-Fender | Take a closer look and double-check your syntax |
| 01:33.34 | josefig | [TK]D-Fender: I want to dial international way like for usa 1+10 digits, Mexico 52+10 digits how should be the proper way ? |
| 01:34.26 | [TK]D-Fender | I just told you one very specific thing about what you just pasted there |
| 01:34.50 | [TK]D-Fender | And you should be looking at that very closely just like I told you to |
| 01:35.26 | josefig | ok |
| 01:37.40 | josefig | [TK]D-Fender: exten=>_Z.,1,Dial(SIP/${EXTEN}@Provider,120,Tt) ; |
| 01:37.44 | josefig | gotcha, thanks |
| 01:41.11 | [TK]D-Fender | Also, that pattern is way too vague |
| 01:41.18 | [TK]D-Fender | That can accept a number like 55. |
| 01:41.24 | [TK]D-Fender | Is 55 dialplan via your provider? |
| 01:41.32 | [TK]D-Fender | How about 704? |
| 01:41.49 | josefig | yes i added now the way how it works |
| 01:42.11 | [TK]D-Fender | "usa 1+10 digits, Mexico 52+10 " <- You should make pattern(s) to better macth numbers that are going to be actually valid to dial |
| 01:42.32 | [TK]D-Fender | exten=>_Z.,1,Dial(SIP/provider/${EXTEN}:2,120,r) <- this will send 33 over if you dial it |
| 01:42.45 | [TK]D-Fender | That doesn't sound like "usa 1+10 digits, Mexico 52+10 " at all |
| 01:44.12 | josefig | ok i understand |
| 01:44.38 | josefig | what's about if I am on Mexico and locally we dial 7 numbers instead of 52 + city code |
| 01:44.53 | josefig | so if we dial 7 numbers i can add a pattern to autocomplete the 52+10 digits format |
| 01:45.04 | [TK]D-Fender | So you make pattens that can match those kinds of number |
| 01:45.07 | josefig | how can add the 52+city code |
| 01:45.19 | [TK]D-Fender | 52 is PART of the pattern |
| 01:45.22 | josefig | yes |
| 01:45.30 | [TK]D-Fender | So make it part |
| 01:46.16 | josefig | but if i dial for example 901-02-03 i can add my city code, such as 222 + 901-02-03 , and the final dial number should be 522229010203 |
| 01:46.19 | [TK]D-Fender | exten => _ <---------make it part of the actual number dialed, and the overall pattern as specific as you can to validate it being a valid & complete number |
| 01:46.29 | josefig | you understand what i am saying ? |
| 01:47.15 | [TK]D-Fender | Yes, so "52" + 10 more digits there |
| 01:47.23 | [TK]D-Fender | So make a patter than does EXACTLY that |
| 01:50.26 | josefig | yes |
| 01:51.29 | josefig | i appreciate it |
| 01:52.09 | [TK]D-Fender | Let's see you show us one just like that now.... |
| 01:54.00 | [TK]D-Fender | Or not... |
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| 03:01.40 | jmordica | Who can recommend a good dedicated hosting company in atlanta or dallas (for VoIP servers obviously)? |
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| 03:17.43 | drmessano | Linode |
| 03:18.34 | jmordica | dedicated servers.. |
| 03:23.49 | drmessano | ok |
| 03:25.03 | drmessano | Rackspace |
| 03:25.43 | drmessano | You can kinda pick them out of google searches |
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| 03:41.33 | snadge | anyone here an expert on php agi? :P |
| 03:42.32 | snadge | im looking at this fastaccounts.agi script.. and its got a $fastagi in it, which isn't defined anywhere.. wtf.. i wonder if thats why its not working |
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| 11:12.18 | Dpunkt | How can i get Asterisk 13.6 with PJsip not to transcode audio? Asterisk takes the endpoint/allow = !all,g722,alaw and would offer g722 to my Phone even if the caller offers only alaw |
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| 13:55.08 | SirLouen | hello! |
| 13:55.37 | SirLouen | anyone can explain me why both private and public keys have to be merged in 1 file in order to use TLS with sip in asterisk? |
| 13:55.59 | SirLouen | is not possible to use them individually? |
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| 15:00.27 | Kunsi | hm, how to check if chan_sip was built with tls support? |
| 15:02.30 | Kunsi | i've set tlsenable=yes and other options, but it seems not to load certificate files (atleast it doesn't show up in cli, like my other installation does |
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| 21:14.03 | SirLouen | hello, second chance: anyone can explain me why both private and public keys have to be merged in 1 file in order to use TLS with sip in asterisk? :-) thanks! |
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