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07:00.23 | fling | Which service is better for outgoing calls to skype? |
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07:34.18 | ChannelZ | What are my choices? |
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09:13.27 | adsc | what is the most frequent problem when a registered SIP phone can't dial out? |
09:13.51 | adsc | my asterisk CLI with verbosity 10 shows nothing when I try to dial out from this phone |
09:14.01 | adsc | dialing in works fine |
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09:18.10 | adsc | here's a paste with my configuration: http://pastie.org/9544377 |
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09:44.41 | adsc | shouldn't I at least see something in the console? |
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09:47.04 | kleszcz | yes |
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09:50.22 | kleszcz | set verbose to 7 and debug to 2 and check again |
09:54.22 | adsc | nothing |
09:54.30 | adsc | i am beginning to suspect the hardphone |
10:06.42 | adsc | yeah, the hard phone was set up to use fixed line for outgoing |
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10:28.23 | wasanzy | hello friends |
10:28.54 | wasanzy | please how can I configure asterisk to accept incoming calls from anywhere? |
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10:55.09 | wasanzy | any help please? |
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11:06.56 | fling | Which modem to buy for a gsm gateway? |
11:07.12 | eirirs_ | us robotics 56k fax modem |
11:07.13 | fling | do I still need e1550 or is it possible to use something more recent? |
11:07.15 | fling | LOL |
11:07.31 | fling | eirirs_: I have this one but it got damaged by lighting one day |
11:07.36 | eirirs_ | :( |
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11:33.32 | wasanzy | please how can I configure asterisk to accept incoming calls from anywhere? |
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13:10.45 | funnymanva | Does anyone have experience with Digium's PSTN GW products like G100, G200, etc? I'm trying to figure out how to set the codec used on the E1 interface (aLaw or uLaw) and I don't see the settings anywhere. |
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13:41.31 | malcolmd | e1 should default to alaw. you shouldn't have to set it. |
13:44.31 | WIMPy | Hmm. Pretty interesting. The only thing you don't have to explicitely configure manually? |
13:52.32 | funnymanva | malcomd thanks, we have really horrible audio so I was hoping it was a codec issue, but maybe it's an encoding issue. They use B8ZS, but it doesn't appear the G400 can support that. |
13:56.38 | [TK]D-Fender | It does |
13:56.44 | [TK]D-Fender | that is absolutely bog standard |
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13:59.32 | funnymanva | For Coding on the E1 it only has AMI and HDB3 as options, I see no selection for B8ZS on the G400 |
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14:12.18 | malcolmd | the e1 equivalent of t1's b8zs is hdb3. you've got a carrier that's requiring/requesting b8zs on an e1 link? |
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14:13.55 | funnymanva | malcomd, we're interfacing to a Nortel PBX (not under our control) and that's what the guy was telling us. I don't think it's an encoding issue or signaling wouldn't be working. I think they're using ulaw on E1 trunks (this they did confirm) instead of alaw which is why the audio sounds horrible (almost unintelligible like we're inside an airplane engine). |
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14:16.42 | malcolmd | phone up the Support guys |
14:18.15 | malcolmd | and make sure you're setting the right value for Use Internal Timing on the T1/E1 General Settings configuration menu |
14:19.00 | funnymanva | ok, thanks |
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14:59.53 | marceloamorim | guys, morning, I`m trying to use the calendar.conf, and there is an error, I don`t know if I`m doing the right way. when I use the CLI> calendar show calendar mygooglcall all the informations is there |
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15:00.39 | marceloamorim | but when I set NoOp(${CALENDAR_EVENT(summary)}, the error: There is no event notification datastore on 'SIP/3000-00000001'! |
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15:05.32 | marceloamorim | should I query the agenda first? |
15:14.05 | marceloamorim | To get that, I need to query the calendar first. Very useful. Thx guys |
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15:53.01 | Stefan27 | Does the sound prompt searching as in https://wiki.asterisk.org/wiki/display/AST/Sound+Prompt+Searching+based+on+Channel+Language really work as the wiki page says? I tried putting Set(CHANNEL(language)=en_test) and then Playback(mooon.wav) but this only works when i have created a file called mooon.wav in the sounds/en directory, not when i create it in the sounds/en/test directory |
15:53.34 | Stefan27 | <PROTECTED> |
15:53.34 | Stefan27 | <PROTECTED> |
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15:53.51 | Stefan27 | works when moon.wav exists directly in en but not when it only exists in en/test |
15:53.55 | [TK]D-Fender | "not when i create it in the sounds/en/test directory" != "en_test" |
15:54.19 | Stefan27 | How does en_test work then? |
15:54.35 | [TK]D-Fender | <PROTECTED> |
15:54.49 | [TK]D-Fender | that underscore doe not magically become a slash |
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15:55.01 | Stefan27 | I thought it would, according to the wiki page i linked |
15:55.13 | Stefan27 | Each channel in Asterisk can be assigned a language by the channel driver. The channel's language code is split, piece by piece (separated by underscores), and used to build paths to look for sound prompts. Asterisk then uses the first file that is found. |
15:55.38 | [TK]D-Fender | not to my awareness |
15:55.53 | [TK]D-Fender | use core debug to confirm |
16:01.47 | Stefan27 | core debug doesn't tell me anythign useful, but it does locate the file in en/ with language set to en_test so it must parse it somehow (there's no folder called en_test nor any file called mooon.wav anywhere but in en/ |
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16:10.00 | WIMPy | The versions remain open. Both for the wiki and what you use. |
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16:10.44 | Stefan27 | I use 12.5 |
16:11.03 | Stefan27 | wiki was updated dec 19 2013 thats all i know |
16:11.34 | WIMPy | If the date is correct, that must be new enough for whatever is documented in the wiki. |
16:13.51 | mjordan | That code hasn't changed from god only knows when. |
16:14.00 | mjordan | I'm not sure the wiki page is correct however. |
16:14.20 | mjordan | Asterisk will do a fallback, in that if you have a sound in en_foo, it will play it - but if that sound doesn't exist, it will check en next |
16:14.49 | mjordan | What I'm not sure of is that it will form a path search from the underscores. |
16:15.02 | mjordan | But I haven't dug deep enough to know for sure. |
16:17.33 | mjordan | looking all the way back to 1.8, it looks like it would not form a sub-directory search. I think the wiki page is probably just wrong. |
16:17.48 | mjordan | Based on the example there, it would instead do the following: |
16:18.22 | mjordan | If the channel language is "en_GB_female_BT", check for that folder first, i.e., sounds/en_GB_female_BT |
16:18.37 | mjordan | If it didn't find the sound, check sounds/en_GB_female |
16:18.45 | mjordan | again, if it didn't find it, check sounds/en_GB |
16:18.49 | mjordan | and finally, check sounds/en |
16:19.24 | mjordan | newtonr: mind making an issue to fix that wiki page? https://wiki.asterisk.org/wiki/display/AST/Sound+Prompt+Searching+based+on+Channel+Language |
16:21.55 | newtonr | for sure |
16:22.02 | newtonr | or, not at all |
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16:34.38 | newtonr | https://issues.asterisk.org/jira/browse/ASTERISK-24319 created |
16:46.26 | Stefan27 | setting language to 'en/test' made it behave almost like i wanted but not quite http://pastebin.com/XKbKseWS |
16:47.01 | Stefan27 | I guess I have to re-make my own sound-file-structure based on how it actually works, thanks for telling me jordan |
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16:49.19 | WIMPy | You might want to add that to the issue newtonr just created. |
16:52.43 | Stefan27 | ok i added a comment |
17:12.56 | Dovid | Is there any way of sending CALLERID(name) as blank? If I Do Set(CALLERID(name)=) then it will be the value of CALLERID(num) which I do NOT want. |
17:14.27 | [TK]D-Fender | Show us |
17:22.41 | newtonr | it may be you want to set CALLERID(all) as blank |
17:22.54 | newtonr | IIRC |
17:23.57 | newtonr | although depending on the path of the call, if its going out of PSTN the provider may overwrite whatever you do |
17:24.02 | newtonr | *out over |
17:27.21 | WIMPy | Apropos callerid: I'd still like to know what exactly CALLERID(ani) is supposed to be. |
17:28.38 | WIMPy | From the description, I expected it to be the cumber of the calling line (as opposed to the callers number), but that doesn't seem to be the case. |
17:29.05 | WIMPy | And if it isn't supposed to be that way, where would we put the calling lines ID? |
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17:49.03 | Penguin | dovid: Does it have to be a null value? Can you set it to UNKNOWN instead? |
17:52.17 | Penguin | If I remember right, when I set my CALLERID(name) to UNKNOWN, I discovered an undocumented feature of my ITSP. It actually caused my outgoing callerid info to be anonymous/restricted. |
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18:01.35 | Dovid | Penguin: I can't. I need to pass the number but not the name. Some carriers (verizon fios and cablevision I believe) if they get something in the cnam field they wont do a cnam dip (to save money). As communicaitons in the US turns over to SIP more and more what you set will go end to end |
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18:23.25 | Dovid | in 11.4 if you do Set(CALLERID(name)=) then it wont sent anything |
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18:23.44 | Dovid | i wonder what changed. and where in the code can i look for this |
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18:28.22 | [TK]D-Fender | Dovid: Show us.... |
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18:28.53 | Dovid | [TK]D-Fender: A copy of both SIP traces? |
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18:29.23 | [TK]D-Fender | * CLI verbose 10, SIP debug. |
18:29.39 | [TK]D-Fender | AND the pee config |
18:29.44 | [TK]D-Fender | peer* |
18:30.43 | Dovid | standby |
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18:46.18 | Penguin | newtonr, mjordan, file, malcolmd |
18:46.33 | file | hm? |
18:47.23 | Penguin | k5673 could use some ban time. |
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18:48.30 | Dovid | [TK]D-Fender: Here are the traces: http://pastebin.com/HGGfXNW8 |
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18:50.35 | [TK]D-Fender | Dovid: There is no name.... looks normal |
18:50.37 | Dovid | with config here" http://pastebin.com/smf1Brzf |
18:50.44 | [TK]D-Fender | Dovid: now go SET one and test |
18:50.47 | Dovid | TK: The first one has no name, the second trace does |
18:51.28 | Dovid | My issue is that in ast 1.8.22.0 if I set Set(CALLERID(name)=) it will be set to ${CALLERID(num)} |
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18:52.21 | [TK]D-Fender | 1.8.22 is 8 revisions out of date |
18:52.21 | WIMPy | Time to get rid of 1.8 then. |
18:52.26 | [TK]D-Fender | ^ |
18:52.28 | [TK]D-Fender | And that |
18:52.40 | Penguin | file: How about a little /mode +b $a:k5673 |
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18:53.15 | Dovid | I know. I assume I can have it patched to act the way I need it? |
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18:53.40 | Dovid | I stick with what works. I woild love to just move but when we find something stable we stay with it |
18:53.50 | WIMPy | That (and a lot of other goodies) is called Asterisk 11. |
18:54.11 | WIMPy | You just told us it doesn't work for you. |
18:54.19 | Dovid | WIMPy: It's a minor issue. |
18:54.33 | Dovid | vs. much larger issues we have had in the past which makes us hesitant to mvoe |
18:55.45 | mjordan | Dovid: which issues in particular have you had with 11 |
18:55.50 | mjordan | As in, which JIRA issues. |
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18:56.03 | mjordan | I'm honestly very curious, as the differences between 1.8 and 11 are not gigantic. |
18:56.20 | mjordan | And, where there were some issues, have mostly been addressed. The vast majority of the time when an issue is filed against 11, it also applies equally to 1.8. |
18:57.20 | *** mode/#asterisk [+b k5673!*@*] by mjordan |
18:58.09 | Penguin | I guess that'll work just as well. |
18:58.25 | Dovid | mjordan: I did not even touch 11 (for my regular job as opposed to my own stuff which is why I am mostly here). We have had many random issues, even in the same version such as in 1.8. in some versions > 1.8.22.0 Asterisk would random;y "forget calls". SIP calls would just disapear. When the RTP stops and the carrier sent a BYE asterisk would just send a 481. So we rolled back to 1.8.22.0. I dont remember them |
18:58.26 | Dovid | <PROTECTED> |
18:58.57 | mjordan | 1.2/1.4 is out of my hands (and everyone around here for that matter). To be frank, that's like blaming Redhat for Fedora 4. |
18:59.22 | mjordan | For 1.8, I'd really like to know what issues those were and have them reported. We take regressions pretty seriously, and try to make sure that when one happens, we get tests written to keep it from happening again |
18:59.33 | mjordan | If there aren't specific issues in JIRA however, there's not much we can do |
18:59.56 | Dovid | mjordan: You are right and I am not blaming you for the issues that were there then. but becase of the "scary unknown" we stay where it works. One of these years I am going to load up 11 on a test box and see how it works |
19:00.05 | Dovid | for my side work I am on 11 and its been great |
19:00.26 | mjordan | that's good to hear; if you do find something please do let us know |
19:00.26 | mjordan | :-) |
19:00.46 | Dovid | matt: I will. geez. so many people helping me here. i am gona owe a lot of beer |
19:00.51 | Dovid | TK: Coming to Astricon? |
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19:01.37 | [TK]D-Fender | Dovid: Nope. |
19:03.32 | Dovid | TK: I dont think you ever came (I think I asked you every year....). it's a lot of fun. especially drunk Matt Riddell |
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19:07.10 | *** mode/#asterisk [+o russellb] by ChanServ |
19:09.55 | Qwell | Dovid: Matt Riddell is the best |
19:11.23 | Dovid | Qwell: Last year he didnt drink though. lets hope this year he is again. no offense to Digium or anyone else but last years party as a tad lame. the year before was fun |
19:11.35 | Dovid | at leat for me |
19:17.18 | Qwell | Dovid: Ask the bartender for 2 drinks at a time. It makes the parties funner. |
19:17.33 | Qwell | Also, find the nearest Snom sales person after the party, and... |
19:17.35 | Qwell | smirks |
19:20.28 | Simon-- | hrm.. seeing an issue with debian wheezy's asterisk 1.8.13.1-based package.. all iax2 peers fail qualify over a minute or so, and won't come back until asterisk is restarted. ping (once per second) is fine the whole time, and the qualify in the opposite direction from remote boxes is still fine. sound like an old bug at all? |
19:21.00 | Simon-- | I set up some stuff to capture iax2 netstats and threads.. |
19:21.50 | Qwell | ~upgrade asterisk |
19:21.51 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
19:23.44 | Simon-- | lols |
19:25.14 | Qwell | Simon--: It was released 22 months ago. It's not worth spending our time trying to figure out if it's already been fixed. |
19:25.48 | Simon-- | sure. I was just wondering if anybody remembers such a bug |
19:26.12 | Simon-- | there are some deadlock-related commits. I will track it down and see if they are related |
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19:28.12 | Qwell | he has returned! |
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19:29.02 | file | yeah, apparently IPv6 routing is a bit ... wonky |
19:29.36 | Simon-- | is there an officialish svn->git mirror? |
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19:37.15 | Qwell | Simon--: tzafrir runs(ran?) one on github I think. I can't google right now, but it shouldn't be too hard to find. |
19:37.31 | Qwell | not official, but it's probably the best you'll find |
19:38.14 | tzafrir | https://gitorious.org/asterisk-tools/asterisk |
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19:56.46 | bananapie | Is there a dial command that would cause newly bridged channels to mute for 5 seconds ( first five seconds ) before connecting the audio ? Transmitting silence would be ok. |
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19:58.04 | newtonr | bananapie, All you have to do is look through the dial help text, though I don't think there is. |
19:58.15 | newtonr | maybe! |
19:58.27 | bananapie | I did look through dial help, I found nothing. A macro might work though. |
19:58.36 | Qwell | macro/gosub should work fine |
19:58.37 | bananapie | wait |
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20:10.50 | Dovid | Snom salesman.... Do i not want to ask? |
20:14.28 | Qwell | Dovid: You really don't. |
20:14.38 | Dovid | will leave it at that |
20:14.53 | Qwell | (there were fighting words) |
20:15.48 | [TK]D-Fender | comes from a place where ALL words are "fighting words" |
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20:22.50 | fullstop | apparently the obihai ATA's now support google voice (again) |
20:23.07 | Qwell | fullstop: officially |
20:23.19 | fullstop | yes |
20:24.03 | fullstop | does this mean anything, at all, for asterisk? |
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20:26.45 | zerohalo | lol - I hope snom salesman doesn't come back |
20:27.01 | Qwell | zerohalo: Were you there?! |
20:27.10 | zerohalo | I was! |
20:27.23 | zerohalo | Saw it all go down. |
20:28.06 | marceloamorim | guys, there is any place too see if ccss(request) is requested? |
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20:42.43 | marceloamorim | like /var/spool/asterisk/outgoing, maybe /var/spool/asterisk/ccss ;) |
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21:14.47 | darkdrgn2k3 | what does [2014-09-11 16:13:21] WARNING[3083][C-0000000e]: channel.c:4860 ast_prod: Prodding channel 'SIP/8999-00000011' failed |
21:14.47 | darkdrgn2k3 | <PROTECTED> |
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