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04:58.21 | jeffspeff | where can i find the table structure for realtime followme and followme_numbers? |
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05:27.21 | igcewieling | Not bad, my VM which does FTP provisioning hit 903 connections when I rebooted a "all" of our hosted phones. |
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12:37.09 | fores331 | Hello, experiencing some hiccups in our asterisk/freepbx installation. Anybody here to help? |
12:44.23 | sibiria | have you tried putting it over your shoulder and patting its back? |
12:45.18 | fores331 | even offered it some tea but still |
12:45.21 | sibiria | if you describe the problem someone who's familiar with freepbx might see and know what's up |
12:49.57 | fores331 | one provider that worked well suddenly got broken and do not allow any outcoming calls anymore but incoming still work or I think, so I am trying to switch to another provider and to most of them I can connect but still can't call. There is one the calls are going through but the voice is interrupted for couple seconds every 2 seconds or so. |
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13:12.20 | jsmith | fores331: Have you looked at the logs? Taken a packet capture of the SIP traffic? |
13:13.16 | fores331 | I tried but don't understand much there |
13:14.50 | jsmith | The first thing I would do would be to try to understand why the provider is rejecting the outbound calls (assuming that they *are* making it to the provider, and that the provider is rejecting them) |
13:15.12 | jsmith | You should be able to look for the outbound SIP INVITE to the provider, and look for its response. |
13:16.30 | jsmith | This may be of some help, depending on your familiarity with Wireshark: https://wiki.wireshark.org/VoIP_calls |
13:19.02 | fores331 | They do it to provider as in their web interface I can see rejection. That provider I am trying to leave and another provider works but with this hiccups. If I connect directly to the trunk with softphone there are no hiccups. So I thought might be some simple settings in freepbx or asterisk sip settings. |
13:23.03 | fores331 | Never seen wireshark but might try to play with it. Otherwise I would be happy if somebody had a look at the server and fixed it for a reasonable fee. |
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15:02.58 | overyander | Hi, I'm setting up pjsip for my provider (Flowroute) that uses IP based authentication. I followed the instructions under "A SIP trunk to your service provider, including outbound registration" at https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples but left out the registration and auth parts. I am able to receive calls from Flowroute but outbound calls fail with https://paste.centos.org/view/c3842b89 what am I missing? |
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15:04.51 | MLC | Could not create dialog to invalid URI 'SECRETaccountkey*155512375555'. |
15:05.03 | MLC | check the specification of the server in your pjsip.conf |
15:05.25 | overyander | specification of the server? |
15:05.44 | MLC | in the aor |
15:05.48 | MLC | contact= |
15:05.51 | file | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
15:06.08 | file | if you are dialing an extension, it's a different format |
15:07.08 | Samot | It needs to be PJSIP/code*${EXTEN}@FLOWROUTE |
15:07.50 | Samot | Or PJSIP/FLOWROUTE/sip:code*${EXTEN}@domain:port |
15:08.02 | Samot | I'd go with the first. |
15:08.42 | Samot | PJSIP/FLOWROUTE/SECRETaccountkey*155512375555 <-- That is a CHAN_SIP format. |
15:09.23 | davlefou | Hi, somes one know what is memberdelay in queue option? |
15:10.32 | MLC | davlefou: look in the sample queues.conf file |
15:10.33 | overyander | thank you file and Samot that solved it. |
15:12.17 | davlefou | MLC good ideas, thk! |
15:13.43 | MLC | Anyone familiar with DPMA? I've added alert= to the phone configuration section, both in the template and directly on the phone itself, but it doesn't show up in "digium_phones show phone xxx". |
15:24.28 | MLC | I got it functional, but the alert= still does not show from the command line "digium_phones show phone xxx". Where do I submit a DPMA bug? |
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15:57.42 | file | Digium phones related things would go through Sangoma Technical Support |
15:57.51 | MLC | ok |
16:03.34 | overyander | I'm having the same issue when trying to dial a local phone extension exten=_XXXX,n,Dial(PJSIP/${EXTEN},30,) |
16:03.48 | overyander | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels says that should work |
16:04.02 | overyander | and I verified that there is a contact for the extension aor |
16:06.19 | [TK]D-Fender | Don't tell us what you verified, SHOW US |
16:06.35 | [TK]D-Fender | Assurances mean nothing |
16:09.00 | overyander | [TK]D-Fender, what do you need to see to help diagnose the issue? |
16:11.32 | [TK]D-Fender | All of it obviously |
16:11.47 | [TK]D-Fender | We have nothing |
16:12.30 | [TK]D-Fender | We have no proof what you're dialing, the state, that it even made an attempt, that your dialplan is even executing what you say, or the kind of failure. |
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16:23.06 | overyander | [TK]D-Fender, how about this? https://gist.github.com/jeffclay/071284690822aac4646925d850b375cb |
16:24.04 | file | that's the dialplan, not the actual console output and what is happening or the "pjsip show endpoints" and "pjsip show contacts" output |
16:24.10 | [TK]D-Fender | Do you should a brochure for your car to a mechanic and then ask what's wrong or do you bring the actual car? |
16:24.41 | overyander | did you guys not see the error and other outputs in that link? |
16:25.11 | file | it did not embed in IRCCloud, but you also need to show "pjsip show endpoint" |
16:25.17 | file | you dial an endpoint, which then references an AOR |
16:25.33 | file | oh and your contact is unavailable |
16:25.44 | [TK]D-Fender | Contact: 1402/sip:1402@192.168.31.27 2e2bc3d7f9 Unavail |
16:25.56 | [TK]D-Fender | Sorry ... nobody home |
16:27.34 | overyander | i've added the endpoint info for 1402 please refresh the page. |
16:27.48 | overyander | what am i missing that is causing my contact to be unavailable? |
16:28.07 | file | unknown, that means that Asterisk sent an OPTIONS SIP request to the given contact and received no response |
16:28.56 | file | is it an inbound registration, or did you explicitly configure it? |
16:29.33 | overyander | inbound reg |
16:29.34 | file | is it a local device, or is it remote? |
16:29.46 | overyander | remote to the server |
16:29.54 | [TK]D-Fender | Look at that IP. |
16:30.02 | file | if it's behind NAT then you need to enable the settings for if a phone is behind NAT |
16:30.09 | [TK]D-Fender | that is not a "local" IP |
16:30.10 | file | rewrite_contact being one |
16:42.00 | overyander | thank you. rewrite contacts did the trick |
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20:44.03 | overyander | can an endpoint have multiple auths? |
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20:47.47 | igcewieling | I doubt any phones support multiple passwords for the same account. |
20:48.55 | igcewieling | Do you have one which does? |
20:58.03 | seanbright | "can an endpoint have multiple auths" yes |
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20:58.32 | seanbright | so sayeth the code |
20:59.15 | overyander | I was thinking more along the lines of having multiple auths for an endpoint to set different usernames AND passwords for an auth ID that is shared by an endpoint. i set my edpoints to be the extension numbers and want to have multiple devices use the same endpoint and aor but have unique authentication. |
20:59.23 | overyander | thanks seanbright |
20:59.27 | seanbright | yw |
21:01.35 | overyander | seanbright, is the auth value in the endpoint config supposed to be comma delimited? |
21:02.40 | seanbright | yes |
21:03.06 | seanbright | looking at the code, it appears that if you specify 'auth' multiple times it will overwrite previous ones, rather than appending |
21:03.11 | seanbright | so comma delimited is the way to go |
21:03.24 | overyander | thank you again |
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21:30.25 | Samot | Oh. Dear. God. |
21:30.54 | Samot | drmessano: I have to deal with, get ready for it, a RED PHONE BOX |
21:32.40 | Samot | Sorry a RedBOX! |
21:33.00 | jpsharp | Odd issue: I have a channel and I set variable __XX=foo. Then I dial out, but in the dial I use a pre-dial handler and change that variable to "notfoo" via an AGI script. I can confirm that the variable is set before the pre-dial returns, but when the channel hangs up, I do a dumpchan and I see the variable twice...once with "foo" and once with "notfoo". |
21:33.08 | jpsharp | I'm very confused. |
21:38.18 | Samot | Man two PRIs that connect to this RedBox that then does T1 signalling to an Adtran TA750 with 16 FXO and FXS ports. Then the RedBox has some special TDM card that is in some ancient FreePBX server, like early 2.x release. |
21:38.24 | jsmith | jpsharp: That sounds like a bug. |
21:39.02 | jpsharp | jsmith: Yeah, I'm digging through the bug tracker it see if it has been reported. |
21:39.45 | Samot | Well it still has Zaptel so yeah, really freaking old. |
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21:49.27 | macaruchi | Hi! |
21:49.57 | macaruchi | I wanna know if you know any GSM gateway that can work with Asterisk without any issue |
21:50.18 | macaruchi | I need a GSM gateaway with at least 16 gsm SIM |
21:50.23 | macaruchi | Any recomendation ? |
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23:27.22 | Slade | think i'll try flowroute for my apps sms verification |
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23:35.41 | Slade | hmm. i guess problem with that is you're seriously rate limited without a shortcode |
23:36.56 | MLC | Twilio does not rate limit on long codes, but their prices are higher. |
23:40.18 | Slade | .004 vs .0075 |
23:40.44 | Samot | Twilio has shortcodes. |
23:41.11 | Slade | Samot, think i just wanted to use a long code on flowroute, but its rate limited |
23:41.35 | MLC | Short codes are so $$$$$ ! |
23:43.11 | Slade | twilio charges .09 if you use their auth service. i figure i can build my own for less than that ;P |
23:43.35 | MLC | yeah, there's not much to it |
23:44.34 | Samot | What is flworoute's rate limit? |
23:44.38 | Slade | trying to find a lib/package out there that does it precanned, so i'm not inventing an already existing wheel.. flowroutes official thing is kinda terribad tho.. uses sqlite db https://github.com/flowroute/sms-verification |
23:46.20 | Slade | 1 per second mms, 5 per second sms |
23:47.33 | Samot | And do you think you're be sending 5 SMS per second for your app verification? |
23:49.53 | Reinhilde | do you think you'll be using MMS? |
23:50.23 | Slade | it's not unreasonable to think that 5 people could try to use it at the same time. but no way it'd be sustained 5per second. but they require you to handle all the throttling and queuing on your side, which makes it a much bigger challenge. |
23:50.45 | Samot | But would they all do it at the same time? |
23:50.54 | Samot | Would all 5 be at 01:00:01 |
23:51.06 | Samot | Or will they be 01:00:01, 01:00:03 |
23:51.08 | Samot | Etc. |
23:51.44 | Slade | given the nature of the app. I think 40-50 people could try within a 1 minute timeframe. giving it a statistical possibility |
23:52.22 | Samot | That's over 1 second per user then. |
23:52.31 | Samot | The rate limit is per second, not per minute. |
23:52.32 | Slade | assuming they all space it out, yes |
23:52.51 | Samot | Do you have this in place now? |
23:52.58 | MLC | A lot depends on how Flowroute measures. If they take an average over in a few seconds you'd be fine |
23:52.59 | Slade | no. i'm implementing it now |
23:53.11 | Slade | MLC, it doesnt. it drops ones over the 5 per second |
23:53.11 | Samot | And how many users will be using this app to start with? |
23:53.20 | MLC | tacky |
23:53.34 | Samot | It's no different then Calls Per Second. |
23:53.38 | Samot | Voice has CPS. |
23:54.02 | MLC | pretty simple to drop the requests into a database or file and have a background process to read them from there and send to FlowRoute |
23:54.20 | MLC | but the additional development cost might not be worth it vs something like Twilio |
23:54.42 | Slade | it's not a large number of users, it's just a small number of users that will all be signing up together at the same time.. anyhow. i guess i can take the time to deal with the rejections |
23:55.03 | Samot | You think you'll have 5 requests being sent in a single second? |
23:55.06 | Slade | MLC, or if i use a precanned system that already does this. i dont have to worry about it ;0 |
23:55.19 | Samot | That's going to be the rarity. |
23:55.35 | MLC | true, but precanned takes some much of the fun out of it! |
23:55.40 | Slade | indeed. it'll happen, but not very often. maybe its just ok to deal with it |
23:55.46 | Slade | MLC, i dont like inventing my own security :P |