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| 02:57.55 | fling | Hello ;> |
| 02:59.22 | fling | exten => ChannelZ,s,Playback(tt-monkeys) |
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| 03:20.52 | mjordan | fling: your syntax is wrong. |
| 03:21.16 | mjordan | fling: syntex should be: exten => {extension},{priority},{application} |
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| 03:34.07 | fling | mjordan: but I have three fields exactly! |
| 03:35.48 | Penguin | But priority s is invalid by itself. |
| 03:36.39 | WIMPy | o.O |
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| 03:41.44 | mjordan | calls it a night |
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| 04:45.43 | ChannelZ | mmwhaa? |
| 04:46.48 | fling | ChannelZ: I'm receiving skype calls via voxeo! :> and calling skype users via zadarma.com |
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| 05:21.24 | madd | hello |
| 05:22.56 | madd | is it possible to include other config files in any *.conf? e.g. sip_department1.conf sip_department2.conf into sip.conf? |
| 05:26.00 | Penguin | Yes. |
| 05:26.27 | Penguin | #include other/files/sip_*.conf |
| 05:26.49 | Penguin | (as an example) |
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| 05:34.14 | ChannelZ | I have no idea what voxeo is but hurray..? |
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| 06:02.43 | newmember | morning |
| 06:03.10 | newmember | just testing an older * system here and I am getting an error on one of the tunks: NOTICE[26552]: chan_sip.c:15432 handle_request_invite: Failed to authenticate user |
| 06:03.39 | newmember | what options should I be looking at to correct this error: NOTICE[26552]: chan_sip.c:15432 handle_request_invite: Failed to authenticate user |
| 06:04.42 | newmember | I have tried messing with 'insecure=' options with no luck |
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| 06:18.12 | Ulf_ | Good morning! |
| 06:19.10 | Ulf_ | I have a bit of a problem with hooking a Cisco SX20 up to Asterisk |
| 06:19.28 | Ulf_ | Would someone be able to hear me out and maybe give some advice? |
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| 06:44.06 | Ulf_ | Okay, is anyone active at all? |
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| 06:48.32 | ChannelZ | A few but no one who knows the SX20 I guess |
| 06:52.54 | Ulf_ | Damn... Oh well, back to Google it is |
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| 07:41.35 | RadJackson | Hi everyone , on our pbx , we may receive calls from customer dialing a SDA (X) or a color number (premium rate number) (Y) , we have just changed the operator , the previous one used to secure the calls , in other words they check if the customer calls Y , if he calls X directly they block the call , with the new operator , they do not have the option , is there any way to scan sip and get |
| 07:41.35 | RadJackson | the called number ? |
| 07:41.45 | RadJackson | I hope my question is clear. |
| 07:42.22 | RadJackson | ${CALLERId(num)} returns the SDA all the time... |
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| 07:51.11 | ChannelZ | You want to know the number they called you at? From most providers that's usually that's the extension. |
| 07:53.12 | RadJackson | The customer should call the premium rate number , this number transfer the call to a DID , this last is routed to my PBX |
| 07:53.29 | RadJackson | Customer should pay for this service , so i do not want them to call the DID directly |
| 07:53.39 | RadJackson | i want to make sure they call the premium rate number |
| 07:54.37 | RadJackson | when i use ${EXTEN} , i get the DID number , even if the customers calls the Premium one, the premium number isnt a physical number ,it is just there to make the customer pay and transfer the call to DID. |
| 07:56.13 | ChannelZ | Is this via SIP or..? |
| 07:56.19 | RadJackson | yes SIP |
| 07:56.57 | ChannelZ | So the first question is whether or not the original dialed number is even provided to you somewhere in the header since they seem to come to the "canonical" extension. |
| 07:57.22 | RadJackson | thats my question , can they send me the origina dialed number thru the header ? i can ask them thought |
| 07:57.31 | RadJackson | Plus how can i scan SIP header when receiving a call ? |
| 07:57.35 | ChannelZ | 'sip set debug on' and then calling yourself will reveal that, look for the INVITE |
| 07:58.04 | ChannelZ | ${DNID} and/or CALLERID(DNID) may or may not give you anything interesting. |
| 07:58.13 | RadJackson | Ok i will check? |
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| 08:06.18 | ChannelZ | I think DNID is mostly for PRI-type interfaces and will likely just show up blank for you. But unless they jam the actually-called number somewhere else in a header, you should probably call the provider and ask them if they can sort their system out to send you calls to the extension of the actual dialed number |
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| 08:09.56 | RadJackson | Exactly , i iwll call the provider and ask them |
| 08:10.00 | RadJackson | thank you ChannelZ |
| 08:11.24 | fling | is e1550 the best modem for a gsm gateway? |
| 08:11.38 | fling | I can buy few here for $7 |
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| 09:11.11 | pa | WIMPy, i tried to contact the author of LCR, more than a week ago now, no answer. Is he supposed to be unresponsive? :) |
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| 10:23.26 | madd | can I set /any/ key=values in [general] or [template](!) e.g. [general] host=dynamic, secret=321, so that the only the "callerid=foobar <007>" have to be set. |
| 10:40.29 | josemslopes | i am trying to play holdtime when some caller on callqueue press 0, so i decided right now to develop in the app_queue.c. (http://pastebin.com/pg2LzunH) |
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| 10:41.48 | josemslopes | i am a newbie on developing on asterisk, so if anyone can guide me on my development, i will thank you. |
| 10:44.00 | josemslopes | the development it is working, but right now i am facing a problem, when a caller press 0 to listen holdtime, and the call is attended by a agent ... the agent only ear the caller, when the playing of holdtime finish. |
| 10:47.21 | josemslopes | it is possible to check when it agent answer the call from the queue for a queue entry, so i can skip the playing of holdtime ? |
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| 11:44.10 | WIMPy | Hi pa. Well, usually no, but he recently got a new full time job, so maybe it has changed. |
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| 11:46.01 | WIMPy | Hi pa. Well, usually no, but he recently got a new full time job, so maybe it has changed. |
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| 12:58.00 | madd | is it a good idea to separate groups in a department via context? And How do I manage phones which in more context than one? |
| 12:58.05 | madd | is that possible? |
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| 13:06.14 | WIMPy | No, but yo can make contexts that include multiple other contexts. |
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| 13:12.06 | madd | WIMPy: e.g. using 2 diff. templates? |
| 13:14.28 | WIMPy | Contexts don't have templates. They have includes. |
| 13:15.21 | WIMPy | Or are you talking about the peers for the phones? In that case you can use templates. |
| 13:17.00 | madd | puh ... i think i haven't the wright terms ... normaly i am not the asterisk guy ... |
| 13:17.16 | madd | i found a working solution for me |
| 13:17.56 | madd | i took numer XXX in two extension groups [default] [spezial] |
| 13:18.11 | madd | and made two templates for each group |
| 13:18.51 | madd | and tell number XXX to use template templ-default and templ-spezial |
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| 13:19.37 | madd | so users of [default] could not call [spezial], and reverse |
| 13:19.37 | WIMPy | I'm not sure what you're talking about, but I have a feeling that you're trying something that won't work. |
| 13:19.55 | WIMPy | And don't use the default context. |
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| 13:21.38 | madd | WIMPy: ok i will rename 'default' to 'something' |
| 13:22.05 | marceloamorim | guys, let me ask something about ccss, there is a way to see the requested? |
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| 13:22.49 | madd | WIMPy: i have phones group1, and also some group2, and one phone which is in both groups |
| 13:22.58 | Qwell | marceloamorim: I think it gets stored in astdb. |
| 13:23.16 | marceloamorim | Qwell: let me check, thx man |
| 13:23.18 | Qwell | file will probably correct me if I'm wrong |
| 13:23.31 | Qwell | I think file was involved in that, anyways... |
| 13:23.57 | file | ummm I was not involved in that |
| 13:24.00 | file | I know nothing of CCSS |
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| 13:24.23 | Qwell | file: putnopvut then! |
| 13:24.27 | file | yes |
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| 13:24.51 | Qwell | file: I always sort of merge CCSS/connectedline in my head, for some reason. |
| 13:25.37 | Qwell | well, I suppose I know the reason, but meh |
| 13:26.08 | marceloamorim | actually, I didn`t find in database show |
| 13:26.35 | Qwell | marceloamorim: maybe ask putnopvut when he shows up later |
| 13:27.00 | marceloamorim | ok, I`ll wait for him, thx guys |
| 13:27.38 | marceloamorim | ccss is lovely tool, it works pretty good, but it helps if we have something to see |
| 13:27.44 | file | I think it's in memory only, not persisted |
| 13:28.22 | [TK]D-Fender | imagine an AstDB problem where it never gets flushed (like a read-only state) and constantly retriggers.... |
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| 13:53.29 | rox | hello, can anyone tell me, what asterisk setting controls the part of PRI SETUP message, that says "Ext: 1Â Progress Description: Calling equipment is non-ISDN. (3)" ? |
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| 14:01.39 | WIMPy | rox: AFAIK that cannot be changed. |
| 14:02.14 | rox | WIMPy: I am having a telco tech telling me, that their line fails connecting some dialed nubers because of that |
| 14:02.22 | rox | demanding that i change that to ISDN |
| 14:02.33 | WIMPy | That doesn't make sense. |
| 14:02.55 | rox | i have googled up contents of SETUP messages, but no indication of what setting would affect that |
| 14:03.02 | WIMPy | Any POTS line or analog port of a PBX would send that. |
| 14:03.04 | rox | WIMPy: it's an albanian branch of vodafone |
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| 14:04.19 | rox | what they did was: they installed an SS7 line and put a SS7<->ISDN-PRI signaling converter there, now they keep demanding from me that i change various SS7 settings |
| 14:06.12 | rox | would the switchtype setting affect that data? |
| 14:06.14 | WIMPy | Sounds like they know what they're doing. |
| 14:06.16 | WIMPy | not. |
| 14:06.31 | WIMPy | Not that I'm aware. |
| 14:06.42 | WIMPy | But I'm away now for 3-4 hours. |
| 14:07.02 | rox | thank you very much for trying |
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| 15:06.52 | rox | How can I set PRI progressmask in an Asterisk/Sangoma system? |
| 15:07.18 | rox | I need to change this: Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] |
| 15:07.21 | rox | into something else |
| 15:07.49 | rox | preferably something that says "This is an ISDN piece of equipment" :-P |
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| 15:17.40 | voip | i got a call today for using asterisk as an sbc / firewall |
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| 15:18.05 | voip | i thought it was interesting they want to terminate l3 sip to asterisk and use asterisk for ser/sbc |
| 15:18.31 | voip | for a pseudo gov facility |
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| 15:32.10 | [TK]D-Fender | voip: * is not an SBC / SIP router. It is a B2BUA |
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| 17:15.30 | gusto | no |
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| 17:16.36 | file | moo |
| 17:17.06 | eirirs_ | moo.c |
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| 17:30.30 | WIMPy | rox: The is no such thing. You can only leave out that IE. |
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| 17:48.26 | Tall-guy | Hey all, Im running 12.5.0 in freepbx 12.0.1rc12....I have inbound from PSTN via PRI-Digium TE133F. I can't get any RINGBACK tones to callers using loadzone=us. Any other loadzone= works fine. (uk/Japan/italy etc). Can anyone shed some light on where to look? |
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| 18:08.07 | WIMPy | Tall-guy: Usually you don't send audio before answering. |
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| 18:28.48 | Tall-guy | WIMPy: these aren't FXO's, these are digital T1's. It's an in-band call progress indicator. |
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| 18:33.16 | newtonr | Tall-guy, since you are using FreePBX for configuration, I'd ask in #freepbx. If it affects only calls coming in over the Digium card, then I'd probably call Digium tech support as well. |
| 18:34.07 | newtonr | Also, yeah my first check would be early audio/progress/ etc, but if it works fine with other loadzone settings, that is strange. |
| 18:34.27 | Tall-guy | newtonr: yeah, the fact that it works with ANYTHING besides US has me stumped |
| 18:35.06 | WIMPy | Tall-guy: Yes, you still usually don't. Or rather can't. |
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| 18:35.08 | Tall-guy | I'm working on a theory, that maybe if "US" is selected, it's expected a certain setting from the Telco provider, and if it's anything but US, it makes a different assumption. |
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| 18:38.28 | [sr] | hi WIMPy |
| 18:38.54 | WIMPy | Hi [sr] |
| 18:42.16 | WIMPy | [TK]D-Fender: You once mentioned a SIP to Up0 gateway. Do you remember a name? |
| 18:45.54 | WIMPy | Found it: Citel |
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| 18:47.48 | [sr] | WIMPy: so whats up?everything normal? |
| 18:48.09 | WIMPy | Define "normal" |
| 18:48.49 | [sr] | WIMPy: never worst than the current :) |
| 18:48.52 | [sr] | got one question, the dahdi configuration for an E1/T1 card, is just like the ISDN BRI one's right? dahdi_genconf and a few more setup and voil+a? |
| 18:49.38 | WIMPy | There isn't much difference between a BRI and a PRI. The number of channels and PRIS are always ptp. |
| 18:50.32 | [sr] | so PRI is ALWAYS PtP |
| 18:50.40 | WIMPy | yes |
| 18:50.53 | [sr] | going to get my hands on the 1st PRI line |
| 18:54.06 | [sr] | well thanks WIMPy for the tip |
| 18:54.26 | [sr] | going to get some rest... i'm up since 6am (20pm now) |
| 18:54.52 | [sr] | its a big sacrifice to wake up before 8am for me |
| 18:55.04 | WIMPy | Sounds evil. |
| 18:55.42 | [sr] | yes :p |
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| 19:01.43 | lnb | http://wiki.freepbx.org/display/L1/TFTP+server |
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| 20:42.22 | Brixius | Hello |
| 20:51.07 | dirty | 180 |
| 20:53.27 | Brixius | I have a few questions on getting asterisk 12.5 and google caldav working, I keep getting Unknown response to CalDAV calendar xxx, request REPORT to /calendar/dav/xxx@gmail.com/events/: Could not read status line: connection was closed by server |
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| 20:54.55 | Brixius | I've built asterisk with neon .30 |
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| 21:06.31 | spicyramen_ | hi people, do you guys know if * 12 supports dtls in trunk configuration ? Im receiving a dtls call from a remote box via this trunk |
| 21:06.41 | spicyramen_ | will test it but want to know if someone already did |
| 21:06.51 | spicyramen_ | this is a pjsip trunk |
| 21:09.02 | file | there are no concepts of trunks in pjsip, they are matched against an endpoint just like a phone |
| 21:09.24 | Brixius | I'm downgrading to neon 0.29.6 to see if that works. |
| 21:10.00 | spicyramen_ | I see @file, maybe thats why FreePbx tag it as "experimental" |
| 21:10.54 | file | the PJSIP work is not yet in a released LTS and it's new |
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| 21:17.15 | spicyramen_ | ok thanks will try chan_sip |
| 21:17.18 | spicyramen_ | thanks mr @file |
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