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| 00:19.17 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 00:19.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.1 (2018/06/11), Standard: 15.4.1 (2018/06/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:30.35 | tuxd00d | What are your thoughts on a solution for this: Local extension(s) ring along with a cell phone, but the cell phone only (not local) needs to confirm acceptance. With FolloMe this works, except if the call rejected (pressing 2), in which case the call is terminated without continuing the dialplan. Dial(SIP/${EXTEN}&LOCAL/${EXTEN_CELL}@delay_5) |
| 00:33.14 | Samot | Solution for what part? |
| 00:33.52 | tuxd00d | Allowing the dial plan to continue after the cell call portion is explicitly rejected. |
| 00:34.06 | Samot | Well they are entering DTMF |
| 00:34.28 | Samot | What do you have for 2 in the IVR context? |
| 00:34.45 | tuxd00d | It's a FollowMe. |
| 00:34.52 | Samot | Right |
| 00:35.05 | Samot | Is the FreePBX? |
| 00:35.48 | tuxd00d | FollowMe(${NameOfFollowMeProfile},dlN) |
| 00:36.04 | tuxd00d | sterisk 15.4.1 |
| 00:36.28 | tuxd00d | It's just a vanilla install. |
| 00:42.09 | Samot | What comes after the FollowMe() call? |
| 00:42.13 | Samot | In the dialplan? |
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| 00:50.29 | tuxd00d | A rejection (press 2), will continue in the "delay_5" context, but it would be nice if it didn't stop the other forks (local extension(s)) from ringing. |
| 00:52.55 | Samot | OK, pastebin the FollowMe profile you are trying to use for this. |
| 00:53.16 | tuxd00d | ~pb |
| 00:53.16 | infobot | methinks pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:54.22 | tuxd00d | well, bummer, the first three pb are gone. |
| 00:54.38 | Samot | pastebin.com |
| 01:00.13 | tuxd00d | The https://pastebin.com/Z6LMbyVn |
| 01:01.15 | tuxd00d | Only things changed are context and number. |
| 01:01.47 | Samot | Show the outgoing context |
| 01:05.24 | tuxd00d | It's a GoSub to the outgoing processing. I can try it directly, but I think the real issues is that FollowMe "answers" the fork regardless of pressing 1 or 2. |
| 01:06.12 | Samot | What fork? |
| 01:06.17 | Samot | I see one number in the profile. |
| 01:06.53 | Samot | Dial(SIP/${EXTEN}&LOCAL/${EXTEN_CELL}@delay_5) <-- However, this...even though it Dials both, which ever one answers will be the winner. |
| 01:07.06 | Samot | The other Dial() channel is dropped. |
| 01:09.34 | Samot | If you're looking to dial multiple destinations, regardless of being local or external, at the same time _and_ have them be handled separately you will need to do multiple Dial() commands. |
| 01:09.36 | tuxd00d | Yes, it would be nice if FollowMe didn't "answer" the channel unless the call is accepted. |
| 01:10.35 | tuxd00d | Oh, I can do multiple dial's separately? |
| 01:10.35 | Samot | You're missing the point here. |
| 01:10.53 | Samot | FollowMe is going to do Dial(EXTEN&CELL) |
| 01:10.58 | Samot | Which is the same as Dial(SIP/${EXTEN}&LOCAL/${EXTEN_CELL}@delay_5) |
| 01:11.04 | Samot | Or close to it. |
| 01:13.04 | tuxd00d | If it were okay, I'd though them all in the FollowMe, but for this application it isn't feasible to "press 1" at local extensions. |
| 01:13.12 | tuxd00d | throw* |
| 01:14.28 | Samot | So, are you putting both the exten and the external number in the followme profile? |
| 01:15.16 | tuxd00d | Not currently, but if the local extensions didn't need to also "press 1", I would just do it that way. |
| 01:15.54 | tuxd00d | Perhaps a compromise of just doing the FollowMe after the locals are done ring might be the way to go on this one. |
| 01:17.20 | Samot | Well if this is during office hours |
| 01:17.31 | Samot | And the user is going to be at the phone most of the time, yes. |
| 01:17.47 | Samot | You should ring the device first and go to followme based the result. |
| 01:18.05 | tuxd00d | Yes, it's for the paranoid owner afraid of missing a call. |
| 01:19.05 | Samot | You could ring the device first, go to FollowMe which rings the mobile and then could ring the mobile and the device again |
| 01:19.07 | tuxd00d | I just didn't like extending the hold time an extra 20 seconds if I didn't have ot. |
| 01:19.32 | Samot | Basically, if they answer a call at the device and it has call confirm message it means they missed the first attempt. |
| 01:19.42 | tuxd00d | Thank you Samot. :) |
| 01:20.13 | Samot | And you can play a "We are locating the user" recording to the caller. |
| 01:20.24 | Samot | After it rings the phone a few times. |
| 01:23.28 | tuxd00d | I know you're not a fan of fail2ban (as a total solution), but I have a friend asking about using it on his small server. The included (with f2b) Asterisk filter doesn't seem to recognize Asterisk 15 Security logs. Do you know where they could find an updated filter (Python RegEx)? |
| 01:45.27 | jpsharp | What's wrong with fail2ban? |
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| 02:41.10 | tuxd00d | jpsharp: It's a reactive solution with a delay. It is be better to put a filter in front of Asterisk. |
| 02:42.11 | jpsharp | Sure, filter out before it gets there, but if your asterisk box is facing the internet and needs to accept SIP traffic from anywhere. |
| 02:51.01 | drmessano | You can set rate limits on REGISTERS and INVITES in iptables |
| 02:51.17 | drmessano | Much more effective than reactively banning a SIP attack |
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| 14:36.39 | nbjoerg | grmbl. why does pjsip insist on using a field named s_addr... |
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| 16:36.42 | sahmed | Hello Good Morning |
| 16:37.28 | sahmed | <PROTECTED> |
| 16:37.43 | sahmed | could anyone please help me about this? |
| 16:39.21 | sahmed | I have tried with SHARED, IMPORT and MASTER_CHANNEL dialplan function |
| 16:40.29 | sahmed | but couldn't manage to read callee part channel variable from an agi which is called on caller channel. |
| 16:41.28 | sahmed | I know there have someone who can help me on this. Its already 3 days I have been struggleing on this. |
| 16:44.07 | rmudgett | https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance |
| 16:46.01 | sahmed | ok let me check this |
| 16:46.49 | rmudgett | Reading your description again you are trying to get information from a callee channel from a caller channel. This is opposite from normal. |
| 16:48.44 | rmudgett | You may want to rethink what you are doing. |
| 16:51.39 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 16:54.52 | sahmed | Let me explain what I want, In my scenario I am using only callee side srtp and I need this info at call end for cdr. So my plan is while call is connected with callee we store this status to a variable and read this at call end |
| 16:55.53 | sahmed | but while we try to read this variable at call end we got nothing, and found this agi was called under caller channel. |
| 16:57.35 | rmudgett | How are you storing this status while the call is in progress? |
| 16:58.51 | sahmed | my plan is to calling a macro from Dial function, like Dial(...,M(abc)) |
| 17:00.03 | sahmed | from this macro we store this with SET function, like, exten => s,n,Set(__SECURE_CALL_CHAN=${CHANNEL(endpoint)}) |
| 17:01.05 | sahmed | but this store the value, but could not read from my agi script |
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| 17:08.26 | sahmed | <@rmudgett> Could you please give some idea? |
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| 17:21.19 | rmudgett | sahmed: You should be able to set the value to a variable on the caller channel: Set(MASTER_CHANNEL(varname)=value) |
| 17:21.38 | rmudgett | On the caller channel you read the variable as normal: ${varname} |
| 17:22.25 | rmudgett | Use MASTER_CHANNEL to access variables on the caller channel from the callee channel. |
| 17:23.28 | sahmed | <@rmudgett> Thanks a lot, let me check this |
| 17:23.30 | rmudgett | The MASTER_CHANNEL accesses the channel identified by the linkedid of the channel doing the accessing. |
| 17:24.20 | rmudgett | If you have local channels in the chain then the MASTER_CHANNEL is going to access the first channel: PJSIP/100 -> Local;1 -- Local;2 --> PJSIP/200 |
| 17:24.52 | sahmed | I haven't any local channel |
| 17:25.26 | rmudgett | PJSIP/100 in this case when PJSIP/200 is accessing the master channel. |
| 17:35.30 | sahmed | using like, exten => s,n,Set(MASTER_CHANNEL(SECURE_CALL_CHAN)=${CHANNEL(endpoint)}) |
| 17:35.51 | sahmed | SECURE_CALL_CHAN is master channels variable |
| 17:36.38 | sahmed | but while I tried to read this at hangup got blank |
| 17:36.41 | sahmed | <PJSIP/srnd0ser11-00000002>AGI Rx << GET VARIABLE "SECURE_CALL_CHAN" <PJSIP/srnd0ser11-00000002>AGI Tx >> 200 result=1 () |
| 17:41.53 | igcewieling | have you tried ${MASTER_CHANNEL(SECURE_CALL_CHAN)} instead of ${SECURE_CALL_CHAN} |
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| 17:42.24 | igcewieling | like: [2018-06-20 13:40:35] <SIP/joy-systems-0000015b>AGI Rx << GET VARIABLE MASTER_CHANNEL(sm_call[1][callid_egress]) |
| 17:42.49 | sahmed | ok let me check |
| 17:46.37 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:48.08 | sawgood | Hi: I have an older Asterisk 1.8 box which has a registred SIP account (working) but when we try to make outbound calls, the other side (Asterisk as well) reports this problem ... |
| 17:48.31 | sawgood | [Jun 20 10:48:09] NOTICE[7833][C-00001684]: acl.c:748 ast_apply_acl: SIP Peer ACL: Rejecting '"client IP' due to a failure to pass ACL '(BASELINE)' |
| 17:48.56 | sawgood | I definitely have the correct DENY and PERMIT statements in sip.conf for the Asterisk 1.8 box ... |
| 17:49.05 | sahmed | <igcewieling> That's work... |
| 17:49.32 | sahmed | <igcewieling> and <@rmudgett> Thanks a lot for your time |
| 17:50.21 | igcewieling | sahmed: I *heart* MASTER_CHANNEL it solves all the stupid variable scope issues. |
| 17:51.17 | sahmed | ok thats great |
| 17:51.35 | igcewieling | sawgood: in many situations the permit/deny don't apply, only host= applies. |
| 17:51.47 | sawgood | I've tried commenting OUT the permit and deny statements, but it still fails |
| 17:51.51 | igcewieling | I think just for peers. |
| 17:52.02 | sawgood | igcewieling: thanks ... I have for host=dynamic ... |
| 17:52.10 | sawgood | since the box registers |
| 17:52.17 | igcewieling | then permit 0.0.0.0/0.0.0.0 |
| 17:52.34 | sawgood | Ok will try that |
| 17:52.55 | sawgood | Its Asterisk 1.8 on the PBX side and Asterisk 13 at the ITSP side |
| 17:53.33 | igcewieling | I've never had good luck with Asterisk's peer matching the way I want, so I allowguest and then handle allowing/rejecting calls as part of my dialplan and AGI |
| 17:53.52 | igcewieling | I do NOT recommend that unless you really know what you are doing |
| 17:54.24 | sawgood | yeah right on |
| 17:54.42 | sawgood | I tried permit=0.0.0.0/0.0.0.0 (and) the same ACL error happens |
| 17:54.59 | sawgood | I've tried having no permit/deny statements as well ... |
| 17:55.23 | igcewieling | if there are not permit/deny statements, then the ones in [general] should apply |
| 17:55.30 | sawgood | looking |
| 17:56.18 | sawgood | The thing is: this new Asterisk box (is just a copy/paste) of the context in sip.conf from other working boxes (and) we only have to change the permit and secret lines to match the box ... |
| 17:56.51 | sawgood | its not like this is a new deal: when a new box comes online, we just copy/paste the context and change the entries to suit ... and this is the 1st time its failed before |
| 17:59.57 | [TK]D-Fender | sawgood, why aren't wel looking at the actual peer & debug? |
| 18:01.26 | sawgood | yeah the INVITE from the Asterisk 1.8 box arrives correctly (but) the ACL error comes up |
| 18:03.01 | [TK]D-Fender | .... |
| 18:03.07 | [TK]D-Fender | peer + debug <- |
| 18:04.34 | sawgood | looking at that now |
| 18:23.36 | sahmed | I am looking for another help, can we set macro in Dial dialplan function along with B or b flag |
| 18:24.24 | sahmed | I want to set b(sub_funct^s^1) and M both with Dial method |
| 18:24.55 | sahmed | can I use them both at a time from Dial? |
| 18:26.24 | sahmed | Actually I need to call a routine at pre-bridge |
| 18:33.54 | igcewieling | Yes, you can use more than one Dial option. |
| 18:36.21 | sahmed | can I call them like Dial(${DIAL_TO},180, b(abc^s^1), M(my_macro))? |
| 18:45.06 | igcewieling | That is a n-00b question, google it. |
| 18:47.21 | sahmed | <igcewieling> thanks, lets googling |
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| 19:02.08 | rmudgett | sahmed: https://blogs.asterisk.org/2017/03/29/dialplan-handler-routines-allow-customization/ |
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| 19:33.34 | nbjoerg | how separate is pjsip from asterisk? |
| 19:34.18 | file | I don't understand the question |
| 19:34.59 | file | you'll need to be specific about what you mean |
| 19:35.00 | [TK]D-Fender | PJSIP is a 3rd party stack. It still needs to be implemented in an application |
| 19:35.06 | [TK]D-Fender | * is one such. |
| 19:35.20 | file | indeed, it's a separate project, Asterisk just has modules which uses it |
| 19:36.36 | nbjoerg | file: there is an annoying misfeature in the pjsip headers, which requires either annoying resorting and hackarounds in the asterisk part or fixing that crap |
| 19:36.41 | nbjoerg | 14:36 < nbjoerg> grmbl. why does pjsip insist on using a field named s_addr... |
| 19:37.31 | file | that doesn't give much context |
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| 19:44.34 | nbjoerg | so pjsip plays #undef games with s_addr, which depends on the include order of the system headers |
| 19:45.04 | nbjoerg | asterisk/network.h is currently typically included *after* the pjsip headers and that in turn gets netinet/in.h and friends |
| 19:45.29 | nbjoerg | at least illumos has some uses in the headers that depend on s_addr to be valid and pjsip breaks that |
| 19:45.59 | nbjoerg | solution 1: fix the crap in the pjsip headers and whereever it is exposed solution 2: reorder all the includes of pjsip headers in asterisk and one more hack in one file |
| 19:46.39 | file | generally if the problem is PJSIP in some way, then it should be fixed there |
| 19:46.54 | file | if the problem is Asterisk, then it should be fixed there |
| 19:47.57 | nbjoerg | yeah, that's kind of why I'm asking how tight the interaction is |
| 19:48.10 | nbjoerg | it can be worked around on the asterisk side, but it is somewhat annoying |
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| 20:53.46 | DannyA | hey guys...im back :-) |
| 20:53.55 | DannyA | so trying to set up asterisk using the FreePBX gui |
| 20:54.06 | DannyA | 1. is that a bad idea? should i just edit the asterisk files manually? |
| 20:54.28 | DannyA | 2. Can anyone help me figure out how to set up my Twilio SIP endpoint as an outgoing trunk? |
| 20:54.44 | DannyA | I spent a few hours late last night and I can't figure out how to set it up, how to set up a simple dialplan to test |
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| 21:05.41 | Dovid | is there any create table out there for sqlite3? https://wiki.asterisk.org/wiki/display/AST/SQLite+Tables |
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| 21:12.15 | DannyA | Samot: drmessano i think it's sending me to voicemail. i have an extension set up for 888, and i thought i have it pointed to the trunk for twilio's sip endpoint |
| 21:12.38 | Samot | No. |
| 21:12.58 | DannyA | when i dial 888 on the softphone it says "please enter ur password followed by the # sign" |
| 21:12.59 | Samot | DannyA: You may have to move your service around at Twilio. |
| 21:13.31 | Samot | DannyA: Programmable SIP is not the same as an Elastic SIP Trunk. That's what you need now. All your devices should connect to Asterisk/FreePBX |
| 21:13.42 | Samot | And then Asterisk/FreePBX connects to Twilio. |
| 21:13.53 | DannyA | yep...i have Zoiper running |
| 21:14.00 | DannyA | connected to the asterisk server running on my pi |
| 21:14.07 | DannyA | just trying to test things out before i move the polycom to it |
| 21:14.23 | DannyA | so i set up a "Trunk" and set the twilio sip domain and u/p |
| 21:14.27 | DannyA | then i set up an outbound route |
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| 21:15.34 | [TK]D-Fender | DannyA, if you're using FreePBX now you should be asking in their channel now, not here |
| 21:15.42 | DannyA | ah! ok! |
| 21:15.46 | [TK]D-Fender | ~freepbx |
| 21:15.46 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:16.27 | DannyA | so one last thing then, my first question above was, should i switch to using just asterisk? |
| 21:16.33 | DannyA | is that an approach you guys would recommend? |
| 21:16.53 | [TK]D-Fender | "only" isn't a thing |
| 21:16.57 | [TK]D-Fender | You want PSN service |
| 21:17.02 | [TK]D-Fender | * doesn't give you service |
| 21:17.10 | [TK]D-Fender | it's a tool you can use WITH a service |
| 21:18.24 | nbjoerg | setting up "pure" asterisk can be quite a bit fun |
| 21:18.30 | nbjoerg | it will also take quite a bit of time |
| 21:18.54 | nbjoerg | whether freepbx is simpler and more efficient depends on a lot of factors... |
| 21:19.54 | salviadud | DannyA, if your setup is simple, you might go for stand-alone asterisk |
| 21:20.54 | DannyA | very very simple set up |
| 21:21.13 | DannyA | i want to be able to dial some numbers, and have it do different things like speed dial (via twilio), curl local scripts (for home autoamtion), stuff like that |
| 21:21.26 | DannyA | i dont need voicemail, intercom (for now!), or any of the bells and whistles that come with freepbx |
| 21:21.54 | DannyA | so right now what i want is: dial 123, it calls my cellphone via twilio, dial 789, it cURL's a local script to turn off my Phillips hue lights |
| 21:22.27 | salviadud | Sounds like a 2 extension dialplan |
| 21:22.38 | salviadud | Yeah, I'd go for stand-alone asterisk |
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| 21:38.06 | Dovid | is there any create table out there for sqlite3? https://wiki.asterisk.org/wiki/display/AST/SQLite+Tables |
| 21:39.22 | [TK]D-Fender | Dovid, not using * to manipulate it |
| 21:39.43 | Dovid | TK: I want to use sqlite3 for real time extensions |
| 21:40.07 | Dovid | seems based on extconfig.conf.sample that it is possible |
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| 21:40.36 | [TK]D-Fender | * doesn't give you any extra tools to manipulate that DB |
| 21:40.41 | [TK]D-Fender | that's up to you |
| 21:42.22 | Dovid | tk: Correct. I just wanted the table structure. from what I understood from the docs sqlite and sqlite3 may have diff table structures. |
| 21:43.27 | [TK]D-Fender | That doesn't make sense |
| 21:43.44 | [TK]D-Fender | * expects fiels where it expects fields.... |
| 21:51.05 | Dovid | [TK]D-Fender: Correct. I need help with creating the table in sql. I understand how I create table is up to me. as per https://wiki.asterisk.org/wiki/display/AST/SQLite+Tables for ast_exten you need to create say commented with a tinyint(1). If I try that it fails. If I just do tinyint then it works so I assume in sqlite3 you dont tell sqlite the size of a column. So I removed all that and it still fails. there |
| 21:51.05 | Dovid | <PROTECTED> |
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