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| 00:17.17 | talntid | pabelanger |
| 00:17.29 | talntid | I have multiple registrations, but I send one to their LV, and another to LA |
| 00:17.41 | talntid | I have not tried multiple to the same location |
| 00:19.15 | pabelanger | talntid, and how do you route inbound DIDs? I can't seem to figure out how to route them based on the sip registration info |
| 00:19.23 | pabelanger | meh |
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| 08:24.20 | tully` | anyone up? |
| 09:07.55 | ChannelZ | nope |
| 09:12.35 | tully` | ChannelZ, is this a ternary condition? |
| 09:12.37 | tully` | exten => s,n,gotoif($["${mmoption}" = "2"]?recmenu) |
| 09:13.09 | tully` | looks like one of those fancy C if else oneliners |
| 09:19.18 | ChannelZ | well, just an if. |
| 09:20.57 | ChannelZ | But yes it's pretty much the same thing |
| 09:21.57 | tully` | kk thanks, a bit new to asterisk so i'm learning as i go |
| 09:23.16 | ChannelZ | Expressions in asterisk get pretty ugly pretty fast |
| 09:27.51 | tully` | haha yea I've noticed. I'm working on a box that has skype for asterisk and am trying to simply route all skype calls to a php script. think i've got it though |
| 09:28.50 | tully` | I'm just checking for a letter in the beginning as no pots phone will transmit one |
| 09:29.06 | tully` | beginning of the CID* |
| 09:30.11 | tully` | eh i guess the proper wording would be the extension or skype name called |
| 09:30.34 | ChannelZ | you can send Skype calls to a specific context |
| 09:31.19 | tully` | well i've got a few special routes that have precedence over this, more of a fallback as the guys actually running this are stupid |
| 09:31.38 | tully` | key skype names that have to do specific functions |
| 09:33.07 | tully` | they add skype accounts to the box and expect them to work immediately, so to prevent countless repitions of the same task i'm just adding a default route for all skype accounts really |
| 09:35.53 | ChannelZ | Not following precisely what you're doing (which direction) |
| 09:37.02 | tully` | yea i'm still a bit green, i've got a few skype accounts that do key functions, like check and make sure each box is recieving skype calls and email if not |
| 09:38.10 | tully` | but the majority of the skype accounts go to the same place and we add accounts as demand grows, so i just rewrote the dialplan to make one default route for skype calls and anything else goes to it's special dialplan |
| 09:39.30 | ChannelZ | you're talking for incoming calls? |
| 09:39.35 | tully` | yea |
| 09:39.40 | tully` | here's a snippet |
| 09:39.41 | tully` | exten => _[a-z].,1,noop(>>>> recieved skype call on ${EXTEN} |
| 09:40.16 | tully` | so any incoming call w/ a letter in the beginning get's routed to the default macro |
| 09:40.44 | tully` | call to an extension with a letter in the beginning* |
| 09:41.00 | tully` | because they're essentially dialing skype names |
| 09:41.10 | tully` | from my understanding at least |
| 09:41.17 | ChannelZ | but what I"m saying is that Skype users (defined in chan_skype.conf) can have their own context. Put them in a 'skype_incoming' context, and you already know it's a Skype call. |
| 09:41.39 | tully` | oh. that seems significantly easier |
| 09:42.28 | ChannelZ | Yeah. |
| 09:42.46 | tully` | lmao thanks, going to rewrite this shit once more |
| 09:43.00 | ChannelZ | heh |
| 09:44.24 | ChannelZ | I guess in your case it's still going to be a wildcard extension to handle new users generically, but it's still a good idea to be handling it in their own context, to differentiate them from "other" calls |
| 09:45.23 | ChannelZ | s/handling it/handling them/ |
| 09:46.22 | ChannelZ | in any event good luck/have fun, I really have to go to bed now |
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| 13:19.27 | gusto | hi |
| 13:19.49 | gusto | does anyone have an idea what protocol is being used on port 5060 / sip ? |
| 13:20.25 | gusto | wikipedia says that it can be anything |
| 13:20.50 | WIMPy | Usually SIP. |
| 13:24.48 | gusto | ??? |
| 13:24.56 | gusto | sip ... what uses sip was my question |
| 13:25.24 | gusto | but however, it worked even without, but i now put it into firewall that he should not block it on both tcp/udp |
| 13:26.05 | gusto | i had to put in rules for rtp anyway ... so ... |
| 13:26.14 | WIMPy | Ah, usually UDP, but it can also be used on TCP. |
| 13:26.35 | gusto | yes |
| 13:26.57 | gusto | aaah, i am done |
| 13:27.03 | gusto | would eat something now |
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| 15:15.12 | wizkid057 | Hello... I'm having a strange issue all of a sudden using a PAP2 Linksys ATA and asterisk... all of a sudden no matter what I do, asterisk silently responds "Unauthorized" to the sip register request... (can only see it with sip debugging on)... I had this issue with another ATA before, but I cant recall how the heck I fixed it... any ideas anyone? |
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| 17:00.13 | bobb_WU | hey there, anyone experienced with gentoo and imap storage for voice mail? |
| 17:00.54 | bobb_WU | apparently my instance was installed from source instead of through portage, and i just upgraded with portage. now imap storage isn't active any longer |
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| 17:38.53 | Foxi352 | Hi once again :-) I try to get video working via SIP. I added videosupport=yes to sip.conf general section, disallowed everything and allowed alaw,ulaw,h263 and h263p in general section and for every video capable sip client. I use x-lite to call the echo test. When clicking on the start video button i get the following negociation: |
| 17:38.54 | Foxi352 | Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) |
| 17:39.11 | Foxi352 | Why does asterisk think it has no video Capability ? |
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| 17:49.31 | [TK]D-Fender | Foxi352: pastebin your actual SIP peer and the complete SIP debug for the call.... |
| 17:51.08 | Foxi352 | [TK]D-Fender: Ok, gove me two mins …. just FYI: re my SIP incoming calls being routed to wrong context for realtime if peer not in cache -> I submitted to jira and it seems in fact that i hit a bug. It has been accepted as unresolved … : https://issues.asterisk.org/jira/browse/ASTERISK-19720 |
| 17:51.15 | Foxi352 | is collecting debug logs |
| 17:59.48 | Foxi352 | [TK]D-Fender: Please: http://pastebin.com/aSt9YgfB I called echo test audio only first, and then pushed the "add video" button. i separated the two sections with a line of "=====================". Sip peer (realtime sql) is attached on bottom. |
| 17:59.50 | *** join/#asterisk irule (~irule@187.139.0.210) |
| 17:59.59 | irule | hi all |
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| 18:02.08 | Foxi352 | [TK]D-Fender: FYI: Asterisk 1.8.11.0-1digium1~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-04-03 14:12:25 UTC |
| 18:15.07 | [TK]D-Fender | "sip show peer 314" |
| 18:16.00 | Foxi352 | ah ok :-) |
| 18:16.46 | Foxi352 | [TK]D-Fender: http://pastebin.com/x3mQjaFu |
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| 18:17.25 | irule | hi my spa3102 registerd with asterisk with sip but when I dial 500 there is nothing moving in the console, am I missing something? |
| 18:17.26 | Foxi352 | hmmm…. i see the codecs line not showing the video codecs ... |
| 18:25.03 | ChannelZ | 2012-04-21 12:24:08 rblsmtpd: 1.246.42.55 pid 29724: FROM: <shittiestu07@realliving.com> BODY=7BIT |
| 18:25.24 | ChannelZ | Nooooo, this is spam!? Address seems legit.. |
| 18:31.25 | jpsharp | You've won $10,000,000 (10 Million United States Dollars)! |
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| 18:33.16 | kresp0 | anyone knows if DUNDi should work on a ipv6 only network? |
| 18:36.20 | [TK]D-Fender | Foxi352: Codecs : 0xc (ulaw|alaw) <------------------------------ |
| 18:36.38 | [TK]D-Fender | Foxi352: Your peer does not have what you thought it should have |
| 18:36.55 | [TK]D-Fender | the "us" part of the negotiation gave that away |
| 18:37.29 | [TK]D-Fender | irule: and are you sitting in * CLI with SIP DEBUG actually enabled and verbose at 10? |
| 18:39.44 | Foxi352 | [TK]D-Fender: I see, do you have any idea why ? Is there something special on * side to add to enable it ? I use X-LITE 4, video enabled and h263 and h263-1998 codecs enabled. And i tried with other softphones too like xmeeting, qutecom e.t.c. |
| 18:39.56 | Foxi352 | IMHO it is not on the softphones side …. ? |
| 18:40.10 | [TK]D-Fender | Foxi352: it is your * PEER |
| 18:41.02 | [TK]D-Fender | Foxi352: You showed a DB dump. I of course do not even see if * is looking at your DB. Or if it is in fact using the DB but is relying on old cahced info and hasn't matched the change |
| 18:41.53 | Foxi352 | it uses the DB because it has no flat config file. But you might be right with the cache ….. good point ... |
| 18:41.57 | Foxi352 | is there any command to flush it ? |
| 18:42.23 | [TK]D-Fender | Probably, but I don't have any experience with that aspect |
| 18:42.50 | Foxi352 | ok, google will again be my friend :-) Thanks for your commitment in the channel here … i might eventually come back to you :-) |
| 18:44.34 | Foxi352 | [TK]D-Fender: FYI: sip prune realtime peer 314 if someone asks again somewhen |
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| 18:45.50 | [TK]D-Fender | Foxi352: Got the result you were looking for now? |
| 18:45.59 | Foxi352 | http://pastebin.com/bVqnDxsv |
| 18:46.10 | Foxi352 | Codecs are showing up in peer dump |
| 18:46.20 | Foxi352 | But videosupport:no is still in the same dump too |
| 18:46.42 | Foxi352 | I will sip debug again to see what cap. negoc. writes |
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| 18:49.47 | Foxi352 | [TK]D-Fender: Ok, negotiation seems better now: http://pastebin.com/kdxA7x37 |
| 18:50.11 | Foxi352 | [TK]D-Fender: But still no video. "Unable to start video" on my x-lite client. I only see my own video, but not the echoed one .. |
| 18:50.56 | [TK]D-Fender | Foxi352: I don't see any dialplan processing there... |
| 18:51.30 | Foxi352 | Want me to enable verbose ? |
| 18:51.46 | [TK]D-Fender | Foxi352: You should never NOT have verbose |
| 18:51.55 | Foxi352 | :-) ok |
| 18:52.10 | Foxi352 | That pastebin was when i pushed the "add video" button while being in an audio call |
| 18:53.05 | bobb_WU | can someone look at my log and help me troubleshoot my odbc problems? pastebin.com/xcygijjL1 |
| 18:54.22 | bobb_WU | i just updated to 1.8.10.1 via portage on a gentoo system- now i'm getting Congestion errors because the database doesn't always respond |
| 18:55.10 | Foxi352 | [TK]D-Fender: Ok, here is a video call (immediatly, without passing from audio only to video with add video) to echo test, with sip debug on peer, and verbose on maximum: http://pastebin.com/jnCNiuqC |
| 19:03.15 | [TK]D-Fender | Foxi352: First, did Echo terminate as fast as it appeared to? |
| 19:03.45 | Foxi352 | No, i terminated it with # after having the video error warning box |
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| 19:05.04 | Foxi352 | there were approx. 5 seconds between the echo and the playback line, means before i terminated with # |
| 19:05.27 | [TK]D-Fender | Foxi352: this is point where you quesiton whether Echo ever supported video in the first place.... |
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| 19:05.56 | [TK]D-Fender | Foxi352: Which I immediately doubt of course. If you want to test Video... do something real with it |
| 19:10.07 | irule | when I recieve a call from trunk, my spa3102 attached phone rings and a few seconds later, my sip phone rings, how may I fix this? |
| 19:10.21 | Foxi352 | Oh …. as i googled for days before asking here, i read about echo test supporting video i *think* starting from 1.4 or so …. And a lot of forum answers to video problems were "try with echo test first" …. but you might be right. I will just boot my mbp and start a second x-lite and try to do a video call between these two …… |
| 19:12.49 | [TK]D-Fender | Foxi352: Excellent idea |
| 19:24.01 | Foxi352 | [TK]D-Fender: Idea was good, but result is the same :-) |
| 19:24.07 | Foxi352 | [TK]D-Fender: Still no video… http://pastebin.com/sBuANLVX |
| 19:24.21 | Foxi352 | On both ends i see local video and error about remote video |
| 19:34.39 | Foxi352 | [TK]D-Fender: I found it. Sorry wasting your time…. Remember i told you in sip show peer 314 the codec showed up but video support was no. I found out that i had videosupport=yes in sip.conf general section, but under the dozen of columns in sip_devices realtime db table there was also a column videosupport per peer …. And it defaulted to no ... |
| 19:34.47 | *** part/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 19:35.19 | Foxi352 | [TK]D-Fender: A big thank you for you patience and devotion. I hope i can challenge you again if i have a problem where google and experience can't help me :-) |
| 19:35.25 | Foxi352 | I really appreciate this |
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| 21:25.52 | AgroTemp | Is there anyone here who has experience with using spandsp? |
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| 22:02.05 | AgroTemp | Is anyone here? |
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| 22:07.44 | pabelanger | ~ask |
| 22:07.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:07.48 | pabelanger | AgroTemp, ^ |
| 22:08.14 | AgroTemp | Oh well, I was looking up tone detection. Do you know how to do it? |
| 22:08.34 | pabelanger | what type of tone detection? |
| 22:08.45 | AgroTemp | Like, I'm writing a C program that will be called on a certain extension of AGI, and I need to get the tone frequency detection |
| 22:08.50 | AgroTemp | 2600mhz for example |
| 22:10.16 | AgroTemp | Are there any libraries or built-in Asterisk features that allow this? |
| 22:10.18 | pabelanger | well, asterisk will detect all the normal telephony tones. EG: dtmf, fax.. anything more then that, you need to add code to asterisk |
| 22:10.43 | AgroTemp | oh |
| 22:11.24 | AgroTemp | Okay, well I found something called mwanalyzer that can take an ast_channel* variable in along with some data and it can find tones. How would I retrieve an ast_channel* variable? |
| 22:13.03 | pabelanger | from what AGI? |
| 22:13.16 | AgroTemp | yeah |
| 22:13.16 | pabelanger | ast_channel is a structure within asterisk |
| 22:13.23 | AgroTemp | I know that |
| 22:13.29 | pabelanger | so if you want to access it, you need to create an asterisk module |
| 22:14.34 | AgroTemp | Well, right now, I'm using a library called CAGI to do commands and stuff. Now I have to create an asterisk module? ._. |
| 22:15.06 | pabelanger | I don't know. You haven't told us what you want to do. |
| 22:15.50 | AgroTemp | Well, I plan to receive sound input from the user and I need to detect if any 2600 mhz frequencies are present in that sound |
| 22:17.49 | pabelanger | Ya, asterisk cannot do that. So you'll need to write your application to handle that. Download the source and start looking at main/indications.c. Or hire an asterisk programmer |
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| 22:58.42 | wizkid057 | Hello... I'm having a strange issue all of a sudden using a PAP2 Linksys ATA and asterisk... all of a sudden no matter what I do, asterisk silently responds "Unauthorized" to the sip register request... (can only see it with sip debugging on)... I had this issue with another ATA before, but I cant recall how the heck I fixed it... any ideas anyone? |
| 22:59.24 | WIMPy | Reverse what broke it. |
| 23:00.01 | wizkid057 | that's less than helpful... |
| 23:00.21 | wizkid057 | quite literally nothing I know of has been changed. I havent had the need to even look at this server for months until I was told this ATA wasn't working. |
| 23:00.39 | WIMPy | Obviousely somethig has changed. |
| 23:01.23 | wizkid057 | no idea what it could be |
| 23:01.48 | wizkid057 | which is why i'm asking here to see if I can figure out what would cause asterisk to silently reject the registration |
| 23:01.55 | wizkid057 | then maybe I have a direction to start in |
| 23:02.25 | WIMPy | I doubt someone here has made that change. |
| 23:02.32 | WIMPy | Maybe someone reset that ATA? |
| 23:02.45 | wizkid057 | i'm not saying anyone has made the change... |
| 23:03.04 | wizkid057 | i'm asking if anyone knows, in general, why asterisk would reject a registration of a device silently |
| 23:03.25 | WIMPy | You're the one most likely to have an idea what's going on. |
| 23:03.25 | wizkid057 | for example, if I change the password on the ATA, the rejection appears in the log and on the console, but when the password is correct the registration rejection is not output anywhere in asterisk except sip debug |
| 23:03.51 | wizkid057 | sighs |
| 23:04.03 | WIMPy | Ok, that's some information. |
| 23:04.20 | WIMPy | But I haven't seen that. |
| 23:04.35 | wizkid057 | its quite strange |
| 23:06.03 | WIMPy | Does the peer have ip restriction? |
| 23:07.08 | wizkid057 | nope, dynamic |
| 23:07.14 | wizkid057 | http://pastebin.com/GKFFXdHf |
| 23:07.42 | wizkid057 | masked out a few things, but, thats what I get in "sip set debug on" |
| 23:09.19 | wizkid057 | sip peer: http://pastebin.com/va1ysCsk |
| 23:10.06 | wizkid057 | not a NAT issue, because the same error occurs without NAT |
| 23:11.18 | wizkid057 | also, probably another dozen ATAs using this server with no issues |
| 23:11.25 | wizkid057 | same peer config |
| 23:11.35 | wizkid057 | not the same user, but, same template |
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