| 00:07.35 | ipengineer | is there a way to match for any priority in the dialplan like you can with exten? exten=>_X.,1,AGI(db-proxy.php) |
| 00:08.00 | ipengineer | basically I dont want to have to set priority 1. I want it to get passed to the script and the script can look at the channel variables set. |
| 00:09.25 | WIMPy | I'm not sure that makes any sense. |
| 00:12.00 | [TK]D-Fender | ipengineer, No |
| 00:12.18 | [TK]D-Fender | ipengineer, You have to number your priorities. that si a STEP NUMBER. it makes no sense to say "any step" |
| 00:12.20 | WIMPy | What kind of "set priority" are you taking about? |
| 00:12.33 | [TK]D-Fender | ipengineer, Because there is no "next" when the one before is "any" |
| 00:12.57 | ipengineer | WIMPy: Right now the request comes in for priority 1 we return the app to run and then it looks for priority 2. We dont want to have to set that for every single option |
| 00:13.07 | [TK]D-Fender | ipengineer, Now you don't NEED to have a priority 1 .... except that in many places where things come in they look for "1" exclusively. |
| 00:13.16 | [TK]D-Fender | ipengineer, Like incoming calls from any device |
| 00:13.32 | WIMPy | Why do you return from your script if you want to continue? |
| 00:13.36 | [TK]D-Fender | ipengineer, But you can ORIGINATE a channel out and target anything yuo want on those parameters |
| 00:14.20 | ipengineer | Ok |
| 00:21.05 | [TK]D-Fender | ipengineer, zap/dahdi looks in "context",s,1 or to a dialed extension instead of "s" where applicable. It will then fall back to [default] as context (HORRIBLE idea) and try to match again. Every other channel driver tends to have a targeted extension, and a specified context. These are ALWAY "step 1" Why would it make sense to be otherwise? |
| 00:22.49 | WIMPy | I also fail to see any scenario where thist couls be of any use. |
| 00:23.25 | [TK]D-Fender | Only idea is to compensate for an administrator mistake in configuration "It should just know, or pick one itself!" |
| 00:23.50 | [TK]D-Fender | Which is not a reasonable policy |
| 00:24.23 | [TK]D-Fender | "I want to program, but not be responsible for the code I write being right. It should figure it out and just work" |
| 00:24.34 | ipengineer | we are trying to write a script to replace switch=>realtime. When a call comes in the channel has what extension it is looking for, the script pulls that from the db using those variable. |
| 00:25.04 | ipengineer | The script tells asterisk to exec an app and appdata it gets from extensions table |
| 00:25.34 | WIMPy | Don't return. Keep the call in the script. |
| 00:25.56 | ipengineer | Ok. |
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| 01:16.16 | __sen | any straightforward way to handle the "has left the conference" announcements? |
| 01:17.17 | WIMPy | Same way. |
| 01:26.48 | __sen | using the h extension, or which? |
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| 01:36.22 | WIMPy | yes |
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| 01:38.37 | __sen | hmm, ${EXTEN} tells me which user it was, i guess i just set that in a different variable earlier in the call, and that will be available during the 'h' extension? |
| 01:39.20 | WIMPy | yes |
| 01:39.53 | WIMPy | But generelly using the channels name would be an obvious choice. |
| 01:44.53 | __sen | yay, all working. thanks :) |
| 01:50.45 | lvlinux | ~help Dial |
| 01:50.49 | lvlinux | woops |
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| 02:20.59 | trewq | Is there a way to send a fax (hylafax) through a voip service like callcenteric or voip.ms? I do not have a fax modem. I spent a good while looking for the answer online but I cound not find it |
| 02:21.51 | WIMPy | Look for T.38 |
| 02:22.53 | trewq | Yes, I looked for t.38 and callcentric supports it.. it is a standard.. It is like saying look up tcp/ip |
| 02:23.34 | WIMPy | That's what you need to send faxex via an ITSP. |
| 02:23.53 | WIMPy | Asterisk can do it. |
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| 02:35.28 | Mango45 | Grandstream's default Local RTP Port is 5004, yet Linksys and Obihai have a port range. Anyone know why? And does Grandstream use only port 5004 or is it 5004 and up? |
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| 02:51.41 | linocisco | very novice question! until what versions of asterisk can be affected by HeartBleed? |
| 02:53.25 | WIMPy | unrelated |
| 02:58.18 | linocisco | WIMPy, until which version, was it still vulnerable for Heartbleed? |
| 02:58.29 | WIMPy | Any |
| 02:58.56 | WIMPy | It doesn't have anything to do with Asterisk versions. |
| 03:02.22 | linocisco | WIMPy, do u mean it is just the flaw of Base Linux OS and its openssl version? |
| 03:02.49 | WIMPy | "just"? |
| 03:02.58 | WIMPy | Yes, it's about ssl. |
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| 05:57.45 | [TK]D-Fender | <Mango45> Grandstream's default Local RTP Port is 5004, yet Linksys and Obihai have a port range. Anyone know why? And does Grandstream use only port 5004 or is it 5004 and up? <- default port only lists a STARTING point |
| 05:58.05 | [TK]D-Fender | Mango45, A single call can take 2 ports per side. If you are on a 3-way call... double that... |
| 05:58.24 | [TK]D-Fender | Mango45, So no, not "only" 5004 |
| 06:04.18 | Mango45 | Thanks. |
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| 06:08.37 | [TK]D-Fender | Mango45, Every device has a range, it's jst a question of if you see it advertised somewhere or as an option. This is a bonus for your networking awareness as far as making sure you don't screw over these important ports as an admin |
| 06:28.11 | linocisco | check the product specific support |
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| 06:39.07 | ruben23 | hi guys anyone can help i got two SIP trunk, but wanted to setup a dialplan..that will autofailover the dialout and incoming calls when my SIP trunk1 will fail or get down |
| 06:39.53 | ruben23 | when sip trunk1 fail all dialout calls wil used the SIPtrunk2, any idea guys how to contract the dialplan |
| 06:41.45 | ChannelZ | Look at ${DIALSTATUS} after the Dial and act accordingly |
| 06:42.06 | ChannelZ | like UNAVAILABLE specifically, if I remember right. |
| 06:43.14 | ChannelZ | CHANUNAVAIL |
| 06:43.18 | ChannelZ | I was sort of close. |
| 06:47.10 | ChannelZ | You might also use the SIPPEER function to look at 'status' ahead of time |
| 06:50.26 | ruben23 | <PROTECTED> |
| 07:02.50 | [TK]D-Fender | Not with all the mistakes in there |
| 07:04.22 | [TK]D-Fender | Line 3 is an invitation to "infinite loop land" |
| 07:04.31 | [TK]D-Fender | You have duplicate labels |
| 07:04.46 | [TK]D-Fender | And you apparently can't spell your variables right consistently. |
| 07:06.00 | ChannelZ | but other than that.. |
| 07:06.13 | [TK]D-Fender | And we also don't see a first dial |
| 07:06.28 | [TK]D-Fender | and as bad as the code I see... I trust the stuff I don't even LESS |
| 07:07.13 | [TK]D-Fender | the IDEA is right. |
| 07:07.37 | [TK]D-Fender | This code is sloppy however. |
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| 07:08.45 | [TK]D-Fender | Oh, and your Gotos are pointless... the place they jump to is THE NEXT LINE... which is it going to fall into anyways |
| 07:09.43 | [TK]D-Fender | Concept : Dial. Still in the dialplan? Check the result. Is this a reason to try another resource? If so continue on to dialing it and repeat the process as far as needed. Otherwise hangup. |
| 07:10.05 | [TK]D-Fender | heads to bed... |
| 07:11.06 | ruben23 | [TK]D-Fender: yeah this dialplan fails.. |
| 07:11.21 | ruben23 | can you suggest somehow.? |
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| 10:49.56 | Rico | hi |
| 10:50.05 | Rico | I have a problem with an 1.8 asterisk |
| 10:50.37 | Rico | Register doesn't work, my asterisk box does not send a new REGISTER with auth after it receives his SIP/2.0 401 Unauthorized |
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| 11:17.26 | Orbixx | I'm trying to set Caller ID in Asterisk call files, but it applies both to the Channel and the outgoing call. |
| 11:18.02 | Orbixx | So if I create a call file that rings me and then rings out to an external phone on some trunk, the set caller ID shows the same on my phone and on the recipients phone. |
| 11:18.40 | Orbixx | Is there a way to have a separate caller ID for the Channel ring and one for the call out to the trunk? |
| 11:24.43 | ipalmer | I have Asterisk 13 setup with realtime pjsip. I have 2 endpoints setup in exactly the same way, one is zoiper, one is a Ploycom phone, when I call zoiper to polycom all ok, but not polycom to zoiper, looking at the SIP logs I can see that the Polycom is transmitting its SIP request to the correct extension but the ip address is our external ip address which I know I haven't configured anywhere on the system, anyone have any |
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| 11:40.19 | Orbixx | How can I dial via a trunk first and wait for an answer, and then ring an internal extension? |
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| 12:35.11 | gavimobile | can someone please explain to me what this means (ChallengeSent) https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_ChallengeSent |
| 12:40.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:41.27 | file | it means a challenge for authentication was sent, it does not constitute an authentication failure |
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| 12:48.29 | gavimobile | file: is this normal asterisk behavior? can't this be a risk? ie I am expiriencing a brute force challenge (challenges), but its not "constituting an authentication failure" how should I prevent this? |
| 12:50.33 | file | it's normal protocol behavior for such things as SIP |
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| 12:52.51 | gavimobile | but isn't this a security risk? |
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| 12:53.23 | gavimobile | I've seen forums online with people having the same issue complaining that they are receiving 1 GB of challenge requests daily |
| 12:53.57 | gavimobile | what is the minimal hd requirement for asterisk anyways? this is excessive stress on the pbx.. shoudltn this be addressed? |
| 12:54.32 | __sen | addressed how? if you have SIP open to the internet, then anyone on the internet can send things to it, and there's nothing asterisk can do about that o.O |
| 12:56.51 | gavimobile | __sen: good point, but the security log in asterisk is showing "asterisk/fail2ban-20141210:[2014-12-09 12:19:36] SECURITY[20649] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1418120376-497592",Severity="Informational",Service="SIP",EventVersion="1",AccountID="fawn",SessionID="0x7fb98042d9a8",LocalAddress="IPV4/UDP/myip/5060",RemoteAddress="IPV4/UDP/37.187.134.27/11924",Challenge="16116a2a"" with a wrong AccountID |
| 12:57.12 | gavimobile | asterisk should have the ability to differentiate between a valid and non valid AccountID |
| 12:57.21 | gavimobile | which ar ethe peers set in the sip.conf file |
| 12:57.28 | gavimobile | are set* |
| 12:57.42 | gavimobile | which the peers are set in the sip.conf file* |
| 12:58.55 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:58.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:00.09 | r00f | it would allow attacker to easily identifiy which users exist in your system, by different reply |
| 13:00.26 | r00f | so asterisk is securing you by asking challenge for all accounts, be it existing or not |
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| 13:02.53 | gavimobile | hrm... |
| 13:03.27 | gavimobile | so what can the asterisk chanel recommend me to do with MB's of exesive logging daily |
| 13:03.53 | gavimobile | or is there any common practive which can be recommended to me ? |
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| 13:06.32 | r00f | change the port so not everyone out there knows you are running asterisk. or limit access to vpn only. or make fail2ban to ban more strictly, so they don't create too much logs |
| 13:12.44 | gavimobile | r00f: that's my problem, I cannot make fail2ban more strict than what it already is because no where does it mention a failure |
| 13:12.58 | gavimobile | it just said accountid='spoofed name' |
| 13:13.05 | gavimobile | but no where does it indicate a failure |
| 13:13.35 | gavimobile | what about trunks that use 5060 |
| 13:13.51 | gavimobile | I can't request them to change ports... |
| 13:13.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:14.12 | gavimobile | I can set a netfilter rule to allow a specific ip to use 5060... mayube that will help |
| 13:14.48 | gavimobile | would that be common practice? |
| 13:15.37 | r00f | i don't know any common practice for task of reducing excessive logging. but filtering it out should help. |
| 13:15.44 | [TK]D-Fender | common enough |
| 13:15.50 | gavimobile | thanks guys |
| 13:19.56 | mjordan | what version of Asterisk are you running? |
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| 13:31.42 | vassilux | Is there a possibility to disable termcap |
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| 13:53.04 | cusco | [TK]D-Fender: I've tried several firmwares, spent quite some time trying and its always the same. Why would asterisk ignore registration requests? |
| 13:53.20 | [TK]D-Fender | cusco: For being malformed in some way |
| 13:53.27 | cusco | ow |
| 13:53.40 | [TK]D-Fender | cusco: What version are you running? |
| 13:54.11 | cusco | right now hold on |
| 13:54.38 | cusco | p0s3-07-4-00 |
| 13:54.45 | cusco | I've tryed a few in http://www.jtech.net/ip_phone/cisco/Cisco_firmwre.aspx |
| 13:54.46 | [TK]D-Fender | of asterisk? |
| 13:54.49 | cusco | ah |
| 13:54.50 | cusco | no |
| 13:54.57 | cusco | asterisk tried in 1.8 and 11 |
| 13:55.18 | [TK]D-Fender | make sure you're on the latest in each branch |
| 13:55.26 | cusco | no I ain't |
| 13:55.29 | cusco | let me check |
| 13:55.40 | cusco | 11.10.0 |
| 13:56.04 | [TK]D-Fender | Once you have I would submit it to the tracker to have a proper dev look at it. |
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| 14:03.47 | Milenco | Hmm, I have a problem where audio drops one-way (the called person can still hear the calling person but not the other way around). this always happens after 15 minutes when my sip provider send a re-INVITE, which we 200OK and receive an ACK back from. |
| 14:03.53 | Milenco | Any ideas what could cause this? |
| 14:05.37 | [TK]D-Fender | Accepting a reinvite across networking that can't support it (typically NAT) |
| 14:05.43 | [TK]D-Fender | reinvites = BAD |
| 14:07.56 | Milenco | is there a way i can fix it within asterisk's sip.conf instead of the firewalls? |
| 14:08.10 | Milenco | i already tried session-timers=refuse to no avail |
| 14:10.23 | [TK]D-Fender | Stop allowing reinvites across places that don't support it |
| 14:12.28 | Milenco | Thanks, I'll dig into it some more :) |
| 14:12.31 | file | there's no way to stop us from accepting a reinvite |
| 14:12.40 | file | you can only stop us from generating them for media redirection purposes |
| 14:14.53 | *** join/#asterisk imox (~imox@p57A97762.dip0.t-ipconnect.de) |
| 14:15.21 | [TK]D-Fender | [09:12]filethere's no way to stop us from accepting a reinvite <- umm... you peer says whether it will or not.... |
| 14:15.30 | file | no |
| 14:15.38 | [TK]D-Fender | oh? |
| 14:15.39 | file | that controls whether we generate a re-invite or not |
| 14:16.04 | [TK]D-Fender | That blows. So we can't turn it down? |
| 14:16.13 | file | making it configurable would break the world |
| 14:16.17 | file | whether to accept it or not |
| 14:16.32 | [TK]D-Fender | gah |
| 14:16.37 | file | if the remote side wants you to send media to elsewhere and you don't then that stream may be broken |
| 14:17.16 | [TK]D-Fender | Milenco: I'd start providing SIP debug for a call to see if anything out of the ordinary is happening. |
| 14:17.22 | file | yeah |
| 14:18.55 | Milenco | I got a pcap already for a bad call |
| 14:19.43 | Milenco | but i cant expect you guys to debug it for me, although i would greatly appreciate it ;) |
| 14:19.53 | *** join/#asterisk Sipster_ (~Sipster@MTLXPQAK-1176264396.sdsl.bell.ca) |
| 14:20.12 | Milenco | Its online at milenco.net/files/.bla/dropped_calls.pcap anyway |
| 14:22.44 | Milenco | this is the related sip.conf: http://pastebin.com/iPapsJjJ |
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| 15:58.10 | tomodachi | what is the status for opus codec support in asterisk? |
| 16:01.58 | Eric-K | In Asterisk 13 with PJSIP, I set 'user_agent=Asterisk 13.0.1' in my pjsip.conf |
| 16:02.09 | Eric-K | Is there a way to replace that with some variable that displays the current version. |
| 16:02.15 | Eric-K | Instead of hardcoding it. |
| 16:08.33 | Milenco | I think I finally solved it file and [TK]D-Fender. I enabled externip and localnet in my sip.conf. That didnt work, but after I also forwarded the SIP and RTP ports to the Asterisk server it seems working now :) |
| 16:09.22 | [TK]D-Fender | Milenco: Well we always saw the invite arriving, so I don't see why any of these would have had an impact... but glad it's working for you now. |
| 16:10.02 | Milenco | Yeah I not too experienced with debugging SIP on this level so I just tried random suggestion I read on the internet |
| 16:10.22 | Milenco | will test some more tomorrow to ensure its working now and to determine the exact cause |
| 16:10.26 | Milenco | thanks for your help tho :) |
| 16:11.31 | [TK]D-Fender | REGISTER that is rather... |
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| 16:16.50 | newtonr | Eric-K, I don't think so. You could probably do something with the System() function and "/usr/sbin/asterisk -V" perhaps? |
| 16:17.11 | Eric-K | I'll play around with that, good idea. |
| 16:17.28 | Eric-K | It's a nice to have, not really that important. |
| 16:17.32 | [TK]D-Fender | Eric-K: #EXEC <- |
| 16:17.52 | Eric-K | Ok |
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| 16:19.31 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Using+The+include+and+exec+Constructs |
| 16:20.10 | newtonr | oh and system is actually a dialplan application and not a function. I haven't used it in ages. |
| 16:20.30 | Eric-K | Exec seems to be something I can work with. |
| 16:20.38 | Eric-K | Reviewing the wiki article. |
| 16:20.54 | newtonr | Yeah I guess system doesn't return anything, other than the result of execution. |
| 16:22.00 | Eric-K | Thanks guys! Appreciate it. |
| 16:22.54 | Qwell | Is there some reason you want remote endpoints to know what version of Asterisk you're running? Good way to let them know you're vulnerable to some exploit. |
| 16:24.38 | Eric-K | Good question! It's just for testing purposes. |
| 16:25.00 | Eric-K | I am testing Asterisk 13 in a sandbox. |
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| 16:46.57 | [TK]D-Fender | [11:20]newtonrYeah I guess system doesn't return anything, other than the result of execution. <- EXEC + GREP = config auto-update |
| 16:47.45 | newtonr | :D |
| 16:47.52 | [TK]D-Fender | Eric-K: And put your useragent in a separate config that gets INCLUDE-d after that exec |
| 16:48.35 | [TK]D-Fender | Eric-K: You could actually just ECHO directly into it when you get down to it, rather than GREP. |
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| 18:01.59 | linocisco | hello |
| 18:02.27 | linocisco | what is the best or mostly used programming language for asterisk? |
| 18:02.53 | scv | for AGI? |
| 18:03.48 | WIMPy | linocisco: You come up with pretty strange questions. |
| 18:04.30 | linocisco | for AGI,AMI or AGI or any new ......to interface tightly with asterisk |
| 18:04.42 | linocisco | for AGI,AMI or ARI or any new ......to interface tightly with asterisk |
| 18:05.49 | *** join/#asterisk monsterco (4632aef5@gateway/web/freenode/ip.70.50.174.245) |
| 18:06.24 | monsterco | Hi everyone - I have a Polycom IP450 that is registers to Asterisk server but I think due to NAT doesn't stay registered. Where are the NAT=YES settings on a Polycom IP450 phone set? |
| 18:06.33 | monsterco | I am using the web gui and not provisioning files |
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| 18:10.00 | WIMPy | linocisco: Whatever you're comfortable with. |
| 18:10.30 | linocisco | is Ruby ok with asterisk? |
| 18:11.03 | WIMPy | linocisco: Asterisk doesn't care what you use. It couldn't find out anyway. |
| 18:14.40 | monsterco | Anyone on Polycom and NAT? |
| 18:16.02 | Katty | A christmas poem, for the asterisk channel. By Katty. |
| 18:16.19 | Katty | Twas the night before asterisk-mas, and all through the hall... not a single phone was ringing, not even a conference call! |
| 18:16.35 | Katty | The auto attendant had been turned off with care, hoping the telemarketers were not be in their hair! |
| 18:16.56 | Katty | The network engineers were nestled all snug in their beds, with visions of time divison multiplexing bounced round their heads. |
| 18:17.32 | Katty | And me with my polycom, and DND mode turned on... Decided to route ALL the calls, to where? Taiwan! |
| 18:18.08 | Mango45 | applauds! |
| 18:19.16 | __sen | :D |
| 18:19.25 | linocisco | WIMPy, ok. i will go ahead with Ruby |
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| 18:27.49 | linocisco | Last Christmas, last year before Christmas, Billy showed how to have fun "happy holidays" on Digium phones using smart BLFs as per http://blogs.digium.com/2013/12/17/happy-holidays-phone/ |
| 18:28.44 | linocisco | this year , what would come up as cool? does it only work for Digium phones? what about other brands? |
| 18:29.16 | drmessano | Last Christmas I gave you my heart |
| 18:29.22 | drmessano | The very next day, you took it away |
| 18:36.05 | linocisco | drmessano, this year, to save from tears, I give it to someone special. Yah Yah, |
| 18:44.59 | Katty | waves to drmessano |
| 18:45.10 | drmessano | waves to Katty |
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| 20:33.43 | am1n0 | hello |
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| 20:35.03 | am1n0 | i'm trying to make an agi script to perform an immediate callback (with a constructed callfile). the callfile gets constructed fine, but how can i move it to outgoing/ *after* the other extension hangs up? the call is hung up to asterisk, but because the line is still busy, the callfile always fails. |
| 20:35.48 | WIMPy | Use the h extension. |
| 20:36.09 | am1n0 | to call another agi script to move the callfile? let me try that |
| 20:36.50 | WIMPy | Or just use System() or one of its siblings. |
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| 20:45.29 | am1n0 | great, it worked. i can't believe i didn't think of that. thanks. |
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| 22:30.38 | Jacoby6000 | How does asterisk query the database on Switch=>Realtime for patterns? For example, searching for exten 707, how does asterisk know to look at _XXX? |
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