| 00:09.45 | apb1963 | returns |
| 00:10.00 | apb1963 | pushes file back into position |
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| 00:22.54 | apb1963 | [TK]D-Fender: ok so I've turned off NAT... still no ring: http://pastebin.com/5KsX6U3i |
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| 00:28.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:37.00 | DMJC | is there an easy way to tell if you are being blocked for outbound calls? |
| 00:37.09 | DMJC | I can indial to my pbx from the mobile network now |
| 00:37.29 | DMJC | I suspect I might be using an account that has outdialling blocked |
| 00:38.11 | [TK]D-Fender | I'd start by looking at how they react to a call. |
| 00:38.21 | [TK]D-Fender | Do you SEE them reject the call? |
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| 00:45.35 | [TK]D-Fender | apb1963, Test calls in the other direction. |
| 00:46.05 | [TK]D-Fender | apb1963, Because from here we see it call, they answer, then it processes dialplan, that device then answers, they bridge. |
| 00:47.51 | apb1963 | [TK]D-Fender: other direction... do you mean reverse the originate details? Or... ?? |
| 00:47.55 | [TK]D-Fender | apb1963, Umm.. something looks VERY wrong there. |
| 00:48.13 | [TK]D-Fender | http://pastebin.com/5KsX6U3i <- |
| 00:48.35 | [TK]D-Fender | Line 7 : Your system starts the callout to your provider - Reliably Transmitting (no NAT) to 66.33.147.150:5060 - INVITE sip:16618365653@66.33.147.150 SIP/2.0 |
| 00:49.20 | [TK]D-Fender | line 199 we see it procewssing dialplan to call out |
| 00:49.22 | [TK]D-Fender | <PROTECTED> |
| 00:50.01 | [TK]D-Fender | line 207 we see ANOTHER callout to that same IP |
| 00:50.04 | [TK]D-Fender | Reliably Transmitting (no NAT) to 66.33.147.150:5060: |
| 00:50.04 | [TK]D-Fender | INVITE sip:66.33.147.150:5060 SIP/2.0 |
| 00:50.07 | [TK]D-Fender | WHY |
| 00:50.13 | [TK]D-Fender | What is in [8007]? |
| 00:50.44 | [TK]D-Fender | You should have GLOBAL SIP debug on here.... |
| 00:50.51 | [TK]D-Fender | This should not be peer-specific |
| 00:50.55 | [TK]D-Fender | like you tried previously |
| 00:55.59 | apb1963 | [TK]D-Fender: [8807] http://pastebin.com/6xF1DcY4 |
| 00:56.14 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 00:56.32 | [TK]D-Fender | apb1963, Well I see ANOTHER callout to them there... |
| 00:56.40 | apb1963 | ?? |
| 00:56.45 | [TK]D-Fender | apb1963, and I don't see debug showing the call to 8807 |
| 00:57.02 | [TK]D-Fender | <[TK]D-Fender> line 207 we see ANOTHER callout to that same IP |
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| 01:01.26 | apb1963 | [TK]D-Fender: http://pastebin.com/tbjAT4pK |
| 01:05.58 | [TK]D-Fender | 339 as soon as you dial out to 8007 it starts ANOTHER call out to your provider |
| 01:06.06 | [TK]D-Fender | You are doing something horribly wrong here |
| 01:07.31 | apb1963 | hangs his head in shame |
| 01:14.14 | apb1963 | wanders aimlessly in circles, looking for the path to redemption. |
| 01:16.05 | apb1963 | hears the dinner bell and starts to salivate |
| 01:17.08 | apb1963 | I need nourishment... can't think straight. A break. |
| 01:17.50 | apb1963 | I will come back later when my brain is hopefully functioning again. Thank you again! |
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| 02:20.33 | Greek-Boy | I'm trying to setup asterisk behind pfsense. I opened all necessary ports. (SIP 5060 and RTP 10000 to 20000). When an external user calls the pbx everything works fine (Echo test, etc) but when users try to call each others extensions no audio. |
| 02:22.32 | [TK]D-Fender | Also make sure you disable port randomization |
| 02:23.07 | [TK]D-Fender | And just because you are forwarding ports doesn't mean you don't ahve the other usual settings to make in [general] and for your respective peers |
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| 11:23.19 | Eric-K | Strange issue with Asterisk 13. When I use Dial(PJSIP/200) there is audio both ways. When I try Dial(PJSIP/200,60,t) there is one-way audio. |
| 11:25.12 | pa | should asterisk say something specific on cli, if one is using a tls connection? i tried to set it up, but i still get Using SIP RTP CoS mark 5 , no mention about SRTP or TLS |
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| 12:26.40 | pa | also a question regarding the SRTP tutorial here: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial#SecureCallingTutorial-Part2%28SRTP%29 |
| 12:27.22 | pa | in the tutorial , we must create the keys also for the client. But some clients do not use that. is it correct? why do we need that stuff also for the client? |
| 12:56.02 | Eric-K | Ah, nvm, found the audio issue. |
| 12:56.11 | Eric-K | nat related setting in pjsip.conf |
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| 14:00.30 | pzn | the verbose asterisk log shows "Registered SIP 5000 at 10.20.30.40 port 5000" for example. however it does not show anything when the register expires. I need this information (sip expiration), how can I see/log this info? |
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| 17:53.17 | ryan_turner | Hi, Im trying to troubleshoot two servers connecting via IAX. Here are the logs and relevant configs. https://gist.githubusercontent.com/ryanturner/ebcdff7bbb181c8dfd8f/raw/5651de27323b1d792c44bd803ae967650ea1a7dd/gistfile1.txt |
| 17:55.50 | [TK]D-Fender | user=memhamwan |
| 17:55.56 | [TK]D-Fender | "user" is not a parameter |
| 17:56.02 | [TK]D-Fender | re-read your sample config |
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| 17:57.08 | Kobaz | hey so... for stuff like this: http://www.telcodata.us/new-exchanges-report?month=1 anyone know the actual data source? |
| 17:57.40 | Kobaz | like that site is run by some guy, but if he disappears off the web, where else would you go? |
| 17:57.45 | ryan_turner | [TK]D-Fender, ah, so it's username? |
| 17:58.05 | Kunsi | Kobaz: Where do you get your data? |
| 17:58.08 | Kunsi | I get it from various sources. Most of it comes from NANPA and the CNAC, as well as various regulatory information pages, CLEC notifications posted on ILEC websites, and user submissions. :-) |
| 17:58.10 | Kunsi | https://www.telcodata.us/faq |
| 17:58.21 | Kobaz | ih |
| 17:58.29 | Kobaz | oh.. did i miss that |
| 17:58.35 | [TK]D-Fender | ryan_turner, Is that what you see in the sample config? |
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| 18:27.31 | chillerz | anybody had any experience with pjSIP realtime using the Kamailio / Opensips database? |
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| 19:41.07 | Eric-K | How can I get an overview of registered pjsip endpoints from the console? |
| 19:41.39 | Eric-K | pjsip show registrations only shows the registration with my carrier but not of my phones |
| 19:46.28 | ChannelZ | pjsip show endpoints |
| 19:51.44 | Eric-K | Thanks ChannelZ ! |
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| 20:14.13 | ChannelZ | Sure |
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