| 00:08.55 | *** join/#asterisk raymondhim (~raymondhi@c-66-41-216-194.hsd1.mn.comcast.net) |
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| 00:11.14 | [Consultant]C-As | stuck on the dial part now... |
| 00:11.39 | [Consultant]C-As | exten => h,n,Originate(Local/18775743418@a2billing2-all-callback,exten,a2billing2-all-callback,5999999999,1) |
| 00:11.39 | [Consultant]C-As | [a2billing2-all-callback] |
| 00:11.39 | [Consultant]C-As | exten => _X.,n,Dial(LOCAL/Voip-Innovations-out/14044905024) |
| 00:12.53 | WIMPy | I don't see any caller ID etting. |
| 00:12.53 | [Consultant]C-As | there is the set callerid with priority 1 i didnt set in context as well |
| 00:13.07 | WIMPy | And do you want to connect two "external" calls? |
| 00:13.33 | [Consultant]C-As | no connect call to IVR |
| 00:14.24 | WIMPy | Those two call legs look very simila except for the number. |
| 00:14.57 | *** join/#asterisk evil_gordita (robert@ip70-188-56-139.rn.hr.cox.net) |
| 00:16.05 | [Consultant]C-As | can i delte one? |
| 00:16.11 | [Consultant]C-As | delete* |
| 00:16.17 | WIMPy | \o/ |
| 00:16.22 | WIMPy | My driver works again. |
| 00:16.50 | WIMPy | You specify two call legs in Originate(). The boths ends of the call. |
| 00:19.21 | [Consultant]C-As | so when originate calls the user, there wont be a callerID |
| 00:19.56 | [Consultant]C-As | or i should say the person getting the call wont see the initial callerid |
| 00:19.56 | WIMPy | No specific one, no. |
| 00:20.23 | [Consultant]C-As | but if they dial out from there i can set callerid right? |
| 00:21.00 | WIMPy | You can set the Caller ID in the dialplan. That's why you would originalte to a local channel. |
| 00:21.31 | [Consultant]C-As | yes , i had that before i think |
| 00:30.14 | [Consultant]C-As | with SIP it actualls dials with local it says it dials but does not :( |
| 00:30.19 | [Consultant]C-As | exten => h,n,Originate(Local/1${callerid}@a2billing-callingcard,exten,a2billing-callingcard,1000,1) |
| 00:31.05 | WIMPy | I've got no idea how you'd integrate with a2b. |
| 00:32.02 | [Consultant]C-As | i havent even passed it too a2billing yet.. that happens after the call back and they select 1 on the ivr |
| 00:32.36 | WIMPy | So what are those contexts you're using there? |
| 00:35.33 | [Consultant]C-As | pretty much a name, i got fed up with trying to get a2billing callback to work, so i did it manually in asterisk.. it was working, just not the callerid.. after the call back it sents it to a2billing context, my ivr answers and if they press 1 it sends them to a2billing agi where they can complete there outgoing calls |
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| 01:23.03 | [Consultant]C-As | after i originate a call after hang up and send to a new context all Variables are lost? |
| 01:25.31 | WIMPy | They gererelly are, unless you use inheritance with _ or __ prefix, but I don't think that would work with Originate(). |
| 01:36.33 | [TK]D-Fender | No such concept |
| 01:36.41 | [TK]D-Fender | When you Originate that is a completely new channel |
| 01:36.46 | [TK]D-Fender | It is nothing like a Dial |
| 01:36.51 | [TK]D-Fender | 100% separate. |
| 01:37.21 | [TK]D-Fender | You can PASS variables to an Originated channel ... but not while using the dialplan app version |
| 01:37.35 | [TK]D-Fender | Only AMI, or call-files |
| 01:37.39 | WIMPy | That's what I expected. |
| 01:37.58 | WIMPy | But you can always place some information in the extension. |
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| 01:44.53 | [TK]D-Fender | Yup, if you want to co about trying to encode it, sure. |
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| 03:44.04 | D30 | hello all, i have some issue with my asterisk.. whenever i try to call from a PSTN to my asterisk box all i can hear is just ringing continously but i can see on my asterisk cli that it has actually picking up the call as i can see the IVR is playing, here is the log > http://pastebin.com/KRtEkM3k |
| 03:47.27 | D30 | how do i start to debug it? |
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| 04:14.34 | D30 | any idea? |
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| 06:10.11 | ChannelZ | D30 what country are you in? Are you set for the right signalling method (KewlStart or not) |
| 06:10.33 | D30 | ChannelZ: thanks for answering.. im from Philippines |
| 06:10.54 | D30 | yes im using kewlstart |
| 06:11.19 | D30 | the pstn line is on channel 1 (port 1) on the card.. |
| 06:12.43 | ChannelZ | Well I don't know why exactly but it sounds like your telco just doesn't know you've answered the line |
| 06:14.31 | D30 | my relevant configuration > http://pastebin.com/tSnEagQE |
| 06:15.44 | D30 | ChannelZ: actually the issue is intermittent, sometimes the call just went through and i can hear our IVR playing.. and sometimes it just keep on ringing while i see our IVR is actually playing (i can see it in asterisk console) |
| 06:16.31 | ChannelZ | What kind of card do you have? |
| 06:16.43 | ChannelZ | oh wait you had the TDM400 |
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| 06:17.20 | D30 | im using the TDM800P ChannelZ |
| 06:17.32 | ChannelZ | Oh. I thought last night you were switching. |
| 06:17.42 | D30 | yes from 400P to 800P |
| 06:17.50 | ChannelZ | Anyway, how many FXO boards do you have on it? Do you have more than one port to try? |
| 06:18.12 | D30 | yes it has 4FXO and 4FXS |
| 06:18.18 | ChannelZ | Make sure it's not a faulty FXO board. Did the TDM400 work presumably? |
| 06:18.45 | D30 | actually thats the reason why im switching cards, because at first the issue is on the 400p card |
| 06:19.00 | D30 | *im thinking the issue is on the 400p card |
| 06:20.32 | ChannelZ | Not sure I'm clear -- did you move FXO cards from the 400 to the 800? Or is it a totally new card with new FXO/FXS boards of its own? |
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| 06:22.35 | D30 | nope i didnt move FXO cards, the 800P has already 4FXO and 4FXS card on it.. |
| 06:22.41 | D30 | its a totally a new card |
| 06:23.37 | D30 | i mean new card, but its not new FXO/FXS card,, |
| 06:23.42 | D30 | i hope you get it.. :) |
| 06:25.29 | ChannelZ | Hmm. Well the only thing I can suggest is check your phone cables and wiring.. it seems like an electrical problem not a software one, assuming Kewlstart is correct for your country/telco (which I don't know) |
| 06:26.46 | D30 | okay ChannelZ , i will double check everything.. |
| 06:27.39 | ChannelZ | have you run the DAHDI timing tests? |
| 06:28.02 | D30 | i did also try to direct the PSTN line to an analog phone and see if it has the same issue |
| 06:28.14 | D30 | ChannelZ: i havent tried the timing test :( |
| 06:28.52 | D30 | searching it now.. |
| 06:29.05 | ChannelZ | run dahdi_test in a shell for a minute |
| 06:29.30 | D30 | ooh okay⦠thanks.. |
| 06:29.35 | D30 | running it now |
| 06:31.40 | ChannelZ | You should pretty much be getting around 99.8% or better |
| 06:32.09 | D30 | yeah 99.998% 99.994% 99.999% 99.999% 99.999% 99.999% 99.999% 99.999% |
| 06:32.36 | ChannelZ | Hmm. |
| 06:32.51 | D30 | does it end? or this will loop continously? |
| 06:33.23 | ChannelZ | it'll just go forever. Hit ctrl-c |
| 06:35.03 | D30 | --- Results after 280 passes --- |
| 06:35.04 | D30 | Best: 99.999% -- Worst: 99.901% -- Average: 99.997766% |
| 06:35.17 | D30 | Cummulative Accuracy (not per pass): 99.998 |
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| 06:38.41 | D30 | i also noticed that when i call my asterisk box, and when it plays the IVR i intentionally hangup my call, i noticed that it didnt hangup, its stil playing the IVR and continously directing the call to an extension.. |
| 06:42.31 | r00f | if both L and S are present in Dial command, asterisk ignores S. is it by design? |
| 07:01.52 | ChannelZ | why use both? |
| 07:02.08 | *** join/#asterisk trurl (trurl@jugando.eth23.net) |
| 07:02.16 | trurl | Hello. |
| 07:02.46 | ChannelZ | greetings earthling |
| 07:03.00 | ChannelZ | Are you getting enough oxygen? |
| 07:07.12 | ChannelZ | guess not |
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| 07:08.52 | trurl | What? Me? Sorry, i had no coffee yet ;) |
| 07:12.57 | D30 | ChannelZ: ive edited/added some in my config http://pastebin.com/DGmdKSzN |
| 07:13.50 | D30 | and also im redirecting the call to an extension not to IVR, seems i havent experience the issue for the past 10minsâ¦. |
| 07:13.52 | ChannelZ | Call progress is a crutch and usually causes more problems than it solves. YMMV |
| 07:14.14 | D30 | hmmnn |
| 07:14.23 | D30 | your suggesting to get rid of it? |
| 07:15.13 | ChannelZ | well let's just say it won't solve your answer problems |
| 07:15.39 | D30 | hmmn okay.. |
| 07:15.50 | ChannelZ | call progress/busy detection/etc is basically making dahdi/asterisk "listen" to the audio and try to provide call progress based on what it hears.. like busy signals. |
| 07:16.25 | ChannelZ | But you might get random hangups when it mistakenly hears someone talking and thinks it's a busy signal. |
| 07:16.58 | D30 | hmmnnn.. |
| 07:17.49 | D30 | so whats the best way to detect hangup signals? |
| 07:18.14 | D30 | i call my box.. it plays the IVR. but when i hangup, the IVR is still playing.. |
| 07:19.39 | ChannelZ | Well if it's your only choice, it's your only choice. Sorry I can't be of much help, I don't know anything about the phone system in your country, if you should have disocnnect supervision or not |
| 07:21.14 | D30 | okay ChannelZ thank you⦠ill try to search some âdisconnect supervisionâ based on PH |
| 07:23.58 | D30 | im experimenting the setting now.. looks it did hangup.. |
| 07:24.18 | D30 | and i havent experience the first issue now.. |
| 07:24.28 | D30 | ChannelZ: thanks for all your inputs.. |
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| 07:40.58 | r00f | ChannelZ: we use a2billing, it uses L to timeout all calls. now, i want to set individual timeout for some trunks, by adding S to the "addtitinal dial string" parameter. so resulting dial string looks like L(...)S(...), but asterisk seems to respect only L |
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| 11:59.22 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 16:08.29 | braden80 | hi folks...looking for some help. just installed asterisk 13, freepbx12 on ubuntu 14.04. having trouble registering my trunk (phonepower) using chan_sip. doesn't seem to be making it through to the provider server, but don't seem to have any dns issues or blocked ports on this side. any ideas? |
| 16:10.58 | braden80 | getting chan_sip.c:4219 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data error, and then Registration for 'XXX@sip.phonepower.com' timed out, trying again (Attempt #57) |
| 16:11.02 | braden80 | anyone out there? |
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| 16:17.52 | [TK]D-Fender | "sip set debug on" <- show us the registration attempt |
| 16:17.53 | [TK]D-Fender | ~pb |
| 16:17.53 | infobot | A "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:17.54 | [TK]D-Fender | ^^^ |
| 16:19.06 | braden80 | i'm seeing the same level of verbosity should i increase |
| 16:19.21 | braden80 | oh wait. i see something. |
| 16:21.09 | braden80 | http://pastebin.ca/2966133 |
| 16:21.21 | braden80 | XXX is my phone # and MY_IP is my external IP |
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| 16:23.22 | [TK]D-Fender | Do not mask this. |
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| 16:23.31 | [TK]D-Fender | we need to see the actul full formatting. |
| 16:25.13 | braden80 | i DM'ed you the link, did you get it |
| 16:25.45 | [TK]D-Fender | looks largely OK. is your server directly on a public IP, or is it NAT'd? |
| 16:27.09 | braden80 | it is behind a router with NAT. settings should be fine i had everything working before using a raspberry pi based asterisk...that kept crapping out so just installed asterisk on ubuntu running on my mac mini |
| 16:27.23 | braden80 | static IP |
| 16:28.34 | braden80 | although maybe i haven't configured this asterisk setup properly for NAT but it looks ok to me |
| 16:36.50 | [TK]D-Fender | Looks OK. make sure the IP in the Contact: we see there is still current and correct |
| 16:36.56 | [TK]D-Fender | and verify your forwarding to your Pi |
| 16:37.05 | [TK]D-Fender | Confirm the ports while you're at it (here) |
| 16:39.01 | braden80 | yes IP is good (static). port forwarding is good too. 5060 UDP & TCP, 10000-20000 UDP all pointing to the internal IP of my asterisk server |
| 16:46.59 | [TK]D-Fender | Ok, so far so good then... you're just not getting an answer from the look of it... |
| 16:47.06 | [TK]D-Fender | verify the host you are registering to. |
| 16:48.46 | braden80 | i'm using the sip.phonepower.com proxy as they give on their credentials page...looks like asterisk is properly resolving to IP |
| 16:49.12 | [TK]D-Fender | it does look that way... |
| 16:49.16 | [TK]D-Fender | What router? |
| 16:49.21 | braden80 | cisco ea2700 |
| 16:49.37 | braden80 | do you think ubuntu is blocking something somehow? |
| 16:49.45 | [TK]D-Fender | It could be... |
| 16:49.52 | [TK]D-Fender | flush your firewall |
| 16:49.54 | [TK]D-Fender | and verify |
| 16:53.15 | braden80 | flushed and confirmed no rules. does that mean everything is blocked or everything is open |
| 16:53.49 | WIMPy | Depends on the policy. |
| 16:54.35 | WIMPy | (=default) |
| 16:54.45 | braden80 | i have gufw running on desktop and it says ufw is OFF |
| 16:56.09 | [TK]D-Fender | that isn't proof |
| 16:56.15 | [TK]D-Fender | iptables --list |
| 17:04.17 | braden80 | hrm. i think i did something. my ssh and vpn conked out |
| 17:07.28 | braden80 | stupid gufw was messing with things. it's off now. |
| 17:08.27 | braden80 | http://pastebin.ca/2966157 |
| 17:08.42 | braden80 | looks like accept for input output and forward |
| 17:08.55 | [TK]D-Fender | ok, and your * debug? |
| 17:09.22 | braden80 | sorry what is that |
| 17:14.15 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 17:16.47 | braden80 | http://pastebin.ca/2966160 |
| 17:16.49 | braden80 | check this out |
| 17:18.43 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 17:24.02 | [TK]D-Fender | [2015-03-29 13:13:32] ERROR[2936]: chan_sip.c:4219 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 17:24.06 | [TK]D-Fender | clear networking issue |
| 17:24.10 | [TK]D-Fender | check your routing, etc |
| 17:25.06 | braden80 | i wonder if i should register the trunk first before any of the phones because they're using the same port. |
| 17:32.59 | [TK]D-Fender | no. |
| 17:33.24 | [TK]D-Fender | you have a core linux networking issue. |
| 17:34.06 | braden80 | hrm. any ideas? |
| 17:34.39 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 17:41.58 | braden80 | for some reason if i set the trunk up using PJSIP it at least reaches the voip provider (even though that doesn't work either) |
| 17:49.39 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
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| 17:53.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 19:36.32 | Kobaz | anyone use panasonic dect/ip phones? trying to get them to auto answer |
| 19:52.10 | *** join/#asterisk v0ip-wolf (~wilfred@2610:1c0:0:2:add4:d8a3:c241:89b2) |
| 19:55.02 | jzu_ | ChannelZ: Yes, it is |
| 19:55.31 | jzu_ | ChannelZ: heh Channel Z... Amiga 1200 ;) |
| 19:56.14 | *** join/#asterisk gusto (~gusto@tmo-106-253.customers.d1-online.com) |
| 20:18.38 | ChannelZ | Well, moreso The B-52's |
| 20:18.53 | ChannelZ | But I was an Amiga person |
| 20:19.48 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 20:23.59 | WIMPy | Asterisk for Amiga? |
| 20:24.44 | jzu_ | ChannelZ: hehe yeah I thought that too =) |
| 20:25.02 | jzu_ | ChannelZ: yes, my Asterisk runs behind NAT |
| 20:25.16 | jzu_ | ChannelZ: phone is in Local network within the Asterisk |
| 20:25.25 | jzu_ | ChannelZ: sometimes I get no dial tone and no audio during the call |
| 20:25.34 | jzu_ | ChannelZ: MOST of the time audio works ok |
| 20:25.58 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 20:26.56 | ChannelZ | Is it a separate machine behind whatever does NAT, or does it run on the same machine? |
| 20:27.28 | jzu_ | its |
| 20:27.41 | jzu_ | doh.. it's Asus router that does the NAT, running MerlinWRT (Linux) |
| 20:28.18 | jzu_ | or well uhm.. yes - I've configured the outgoing trunks to nat=force_rport,comedia also |
| 20:28.44 | ChannelZ | So you need to properly set externaddr and localnet in your sip.conf. You need to port-forward a range of ports, as set in rtp.conf, plus 5060 to your asterisk box |
| 20:28.54 | jzu_ | and I've done port forwards from WAN to Asterisk on UDP 5060 & 10000-20000 |
| 20:29.13 | jzu_ | yes, done all those and still scratching my head :) |
| 20:30.04 | jzu_ | I get retransmission errors sometimes, is that normal with UDP? |
| 20:30.37 | jzu_ | "Retransmission timeout reached on transmission 003094c3-139c001b-1c1b2519-083135e6@192.168.1.8 for seqno 102" dadadaa |
| 20:30.39 | *** join/#asterisk rogerh (~rogerh@202.37.70.18) |
| 20:30.41 | ChannelZ | UDP isn't guaranteed so the factual answer is "yes", but it should be rare |
| 20:30.57 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:31.03 | jzu_ | it got rarer when I enabled multicast routing on the WAN router |
| 20:31.35 | ChannelZ | Asterisk sending retransmissions means one of two things (or possibly both); 1. Your traffic isn't making it past your firewall out to wherever it was going; 2. It did, but the replies never made it back through your firewall to you |
| 20:32.53 | jzu_ | alright |
| 20:34.01 | jzu_ | should SIP Trunk CID overwrite this: From: "100" <sip:100@192.168.1.7> |
| 20:35.23 | ChannelZ | Depends |
| 20:36.13 | jzu_ | I haven't been able to dial out as 35894247055 ever, it always goes as 100 |
| 20:36.25 | ChannelZ | 100 is the ID of the phone you're calling from, yes? |
| 20:37.32 | Penguin | Set() your CALLERID(num) before you Dial(). |
| 20:37.36 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:38.00 | ChannelZ | So when your dialplan just does a Dial(SIP/external/${EXTEN}) or whatever, the CID of your device gets passed through. If you were to Answer() the channel first and then Dial(), asterisk is creating a new call its self and will be the CID set for the peer. |
| 20:38.06 | jzu_ | ChannelZ: yes |
| 20:38.14 | ChannelZ | For the former, as Penguin says, set the CID yourself |
| 20:38.34 | ChannelZ | (there's also a Dial argument for it -- f(number) |
| 20:38.45 | Penguin | In both cases, set it before Dial(). |
| 20:39.06 | jzu_ | From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as141214be <-- 192.168.1.7 is my Asterisk |
| 20:39.11 | WIMPy | Huh? It doesn't matter if you Answer() or not. |
| 20:39.13 | jzu_ | so it doesn't pass the CID at all :) |
| 20:39.50 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:40.13 | jzu_ | is there some place where I should set it in FreePBX or do I need to edit config file by hand? |
| 20:40.29 | Penguin | You may also want to set sendrpid=yes in the peer entry for your ITSP. |
| 20:40.42 | jzu_ | I've tried setting Outbound CID on the Extension, Trunk, Outbound route etc - didn't resolve it |
| 20:40.56 | Penguin | We don't support FreePBX here, so you'll be lucky to get a good answer about that. |
| 20:40.57 | WIMPy | Or maybe more likely today, sendrpid=pai. |
| 20:41.05 | jzu_ | Penguin: hah, got channels mixed up :) |
| 20:42.12 | jzu_ | ahh, yes it must be sendripd that I'm missing |
| 20:42.47 | gusto | hi Penguin |
| 20:42.48 | jzu_ | because well, even with ITSP peer having CID defined it doesn't get passed to ITSP.. |
| 20:42.58 | gusto | Penguin, how are your truck mirrors? |
| 20:43.19 | WIMPy | Callerid setting in the peer is for incomming calls. |
| 20:43.25 | gusto | hi WIMPy |
| 20:43.37 | gusto | WIMPy, what's new? |
| 20:44.02 | jzu_ | WIMPy: ah :) |
| 20:44.02 | Penguin | The callerid value for the peer is what is attached on calls that the peer sends through asterisk. |
| 20:44.32 | Penguin | So if my phone has callerid=1234, calls I make from the phone through asterisk have a callerid value of 1234 on them. |
| 20:44.55 | WIMPy | gusto: You're back. |
| 20:45.03 | Penguin | Do not set callerid= in the peer entry for the ITSP. |
| 20:45.07 | gusto | WIMPy, yes, I moved |
| 20:45.20 | Penguin | Set the callerid value in the dialplan before the Dial(). |
| 20:45.39 | Penguin | There is no other way. |
| 20:45.43 | gusto | WIMPy, now here I can not get Kabeldeutschland cable, so I am considering DSL again |
| 20:46.06 | jzu_ | Penguin: seems that's how FreePBX (sorry;) does it indeed, it's in the dialplan |
| 20:46.23 | gusto | WIMPy, and in the meantime 3G |
| 20:46.29 | WIMPy | Off course you can set the callerid on the originating phone or its peer entry. |
| 20:46.42 | jzu_ | exten => 0942450755,1,Set(__FROM_DID=${EXTEN}) etc |
| 20:46.55 | Penguin | That means nothing. |
| 20:47.25 | Penguin | FROM_DID is an arbitrary variable that isn't relative to the callerid value. |
| 20:50.09 | Penguin | If you don't see Set(CALLERID()=), it isn't setting the callerid. |
| 20:50.50 | *** join/#asterisk bkruse (~Adium@64.89.97.100) |
| 20:55.24 | jzu_ | ah, ok |
| 21:06.55 | jzu_ | Remote-Party-ID: "0942450755" <sip:0942450755@jus.si>;party=calling;privacy=off;screen=no |
| 21:07.08 | jzu_ | well isn't that something that I have been missing? |
| 21:07.37 | Penguin | Has that always been there? |
| 21:09.32 | jzu_ | no, it appeared there now that I set it as Outbound CallerID on ITSP Trunk and did Force Trunk CID on FreePBX |
| 21:09.45 | jzu_ | but still it's showing No Caller when I try to call my mobile, mmmm |
| 21:09.53 | jzu_ | extensions_additional.conf:exten => _0Z.,n,ExecIf($["${KEEPCID}"!="TRUE" & ${LEN(${DB(AMPUSER/${AMPUSER}/outboundcid)})}=0 & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE="0942450755" <0942450755>)) |
| 21:10.00 | jzu_ | seems like FreePBX does something like that to the dialplan |
| 21:10.42 | jzu_ | well that's not what it did, that's done by the outbound route |
| 21:10.49 | jzu_ | dialplan so to say |
| 21:11.22 | jzu_ | uh, I better stop using FreePBX :) |
| 21:11.46 | Penguin | Increase core verbose before testing the call. |
| 21:11.58 | Penguin | It will show you each step of the dialplan as it executes. |
| 21:12.08 | Penguin | If there is a Set(), you should see it. |
| 21:13.32 | jzu_ | http://pastebin.com/cuGCUZsb |
| 21:14.22 | jzu_ | ExecIf("SIP/100-0000007f", "1?Set(CALLERID(all)=0942450755)") in new stack |
| 21:15.14 | Penguin | It is being set three times@ |
| 21:15.28 | Penguin | Once is enough, so three times should be even better... right? |
| 21:15.38 | jzu_ | hah, you'd think so yes? |
| 21:15.43 | jzu_ | but you never know :O |
| 21:16.08 | jzu_ | well then it MIGHT be in the wrong format for Sonetel, I must try 358942450755 etc |
| 21:16.26 | jzu_ | 09blahblah is local 3589 is international format |
| 21:17.19 | jzu_ | should sendrpid=yes be enough or should I try =pai |
| 21:17.46 | Penguin | yes should be fine. I've always used yes and never even heard of the pai setting before. |
| 21:19.19 | WIMPy | It would send a PAI header insetd of an RPID header. |
| 21:19.46 | Penguin | Perhaps if PAI is required, or if RPID is ignored... |
| 21:19.59 | Penguin | I just always used yes. |
| 21:20.03 | WIMPy | I haven't seen anyone using RPID. It's all PAI for me (so far). |
| 21:20.40 | Penguin | I think I've only used US ITSPs. |
| 21:20.43 | jzu_ | is there seperate option for PAI or do you mean sendrpid=pai ? |
| 21:22.50 | WIMPy | sendrpid=pai |
| 21:23.12 | WIMPy | It's te same for trustrpid. That can also be set to pai instead. |
| 21:23.34 | Penguin | I've always used trustrpid=yes as well. |
| 21:25.54 | jzu_ | still no CID on my mobile : ) |
| 21:26.22 | Penguin | Check to see if there is a setting on the provider's side that you need to enable or change. |
| 21:27.56 | Penguin | You tried sendrpid=yes and then changed it to sendrpid=pai and it still doesn't work? Are you reloading after each adjustment so settings are applied? |
| 21:29.05 | jzu_ | Penguin: yes and yes |
| 21:42.51 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 21:43.25 | jzu_ | oh well, I'll deal with the CID later |
| 21:43.48 | jzu_ | worse problem is that sometimes outgoing calls don't have dialtone at all and line doesn't have audio |
| 21:55.34 | WIMPy | The dialtone is provided by your terminal, long befor it tries to communicate with Asterisk. |
| 22:02.45 | *** join/#asterisk braden80 (4c0aad7a@gateway/web/freenode/ip.76.10.173.122) |
| 22:04.23 | braden80 | anyone here using phonepower? need some help configuring asterisk with PJSIP |
| 22:04.35 | braden80 | got inbound working, but outbound fails |
| 22:05.42 | ChannelZ | I think he might be confusing dialtone with ringing/progress tone |
| 22:06.12 | ChannelZ | braden80, fails how? Show us some console output, or pjsip debug.. |
| 22:06.28 | ChannelZ | (pjsip logger) |
| 22:09.53 | braden80 | here is on verbose & debug level 10 as i attempt the call |
| 22:09.54 | braden80 | http://pastebin.ca/2966275 |
| 22:11.02 | ChannelZ | ugh. that's debug-debug. Way too much |
| 22:11.07 | braden80 | lol :) |
| 22:11.11 | braden80 | k what level should i change it to |
| 22:11.17 | ChannelZ | zero |
| 22:11.19 | ChannelZ | no debug |
| 22:11.27 | ChannelZ | core set verbose 3 |
| 22:11.31 | ChannelZ | pjsip set logger on |
| 22:11.53 | braden80 | k |
| 22:14.02 | braden80 | http://pastebin.ca/2966277 |
| 22:14.25 | braden80 | starting line 27 i think |
| 22:17.07 | ChannelZ | ugh.. freepbx again.. |
| 22:17.21 | ChannelZ | so phonepower is rejecting you, for whatever reason |
| 22:17.26 | braden80 | yeah |
| 22:17.40 | braden80 | incoming calls working ok. i think it might have something to do with the PJSIP registration |
| 22:19.24 | ChannelZ | Well they never even request auth, so they don't know who you are for some other reason.. |
| 22:19.48 | ChannelZ | trying to dig through their support to see what their requirements are |
| 22:20.03 | braden80 | their support is minimal at best |
| 22:22.50 | ChannelZ | Wellllll, I see something about "activating SIP credentials" in your account, to which they also have these caveats about how many concurrent calls you can have, etc. But I don't see any explanation of what happens without this turned on.. |
| 22:23.26 | braden80 | i do have that turned on and it provided me with basic setup information (sip server, username and password) that i've used to configure the PJSIP |
| 22:25.06 | braden80 | it shows that i'm registered in asterisk and in the phonepower website it shows me as registered. incoming calls work fine, just can't place outgoing calls |
| 22:28.21 | ChannelZ | well I'm not sure how much I can help since on top of this you're using freepbx |
| 22:29.18 | ChannelZ | I'd say 'show me your pjsip.conf, XXX out your username and YYY your password' |
| 22:29.38 | braden80 | i can show you |
| 22:33.06 | *** join/#asterisk monsterco (463336af@gateway/web/freenode/ip.70.51.54.175) |
| 22:34.10 | monsterco | Hi everyone - I have a Linksys WRP400 (which is the ATA + Router) and when I do a SIP registeration on Line 1 and make calls it seems like my fax is dialing only 47 out of a number like 4164792323 - or maybe the dialplan on ATA is not good (which is what I am suspecting) |
| 22:34.28 | monsterco | Can anyone write their standard north american dialplan for linksys ATA? thanks |
| 22:36.49 | braden80 | ChannelZ: http://pastebin.ca/2966280 |
| 22:36.51 | ChannelZ | ([2-9]xx[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx|911) |
| 22:37.51 | braden80 | (*xx|xxx|[2-9]xx[2-9]xxxxxx|1 [2-9]xx[2-9]xxxxxx) is what i have |
| 22:38.09 | braden80 | to allow for *XX and any 3 digit call (311/411/911) |
| 22:39.47 | braden80 | monsterco: 416 ? :) From toronto too? |
| 22:42.02 | monsterco | braden80 - yes, TO :) - I think I have that and for strange reason it's not working |
| 22:42.26 | monsterco | You have a space in dialplan - did you copy paste this? |
| 22:42.40 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 22:42.55 | braden80 | who me? i did copy paste...it's always worked for me |
| 22:43.03 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 22:43.06 | monsterco | I think *XX for feature codes and xxx for three digits |
| 22:43.27 | monsterco | (*xx|xxx|[2-9]xx[2-9]xxxxxx|1*SPACE HERE*[2-9]xx[2-9]xxxxxx) |
| 22:43.54 | braden80 | you can use spaces for clarity it doesn't affect the dialplan |
| 22:44.36 | braden80 | you can take it out if you want |
| 22:45.15 | braden80 | monsterco, maybe your asterisk server is doing something to your dialplan before it dials out (like prepend or replace) |
| 22:46.50 | *** join/#asterisk JosephP (~joe@33.14.200.146.dyn.plus.net) |
| 22:49.39 | monsterco | I see 47 dialed as the LAST NUMBER on ATA so I am guessing this has nothing to do with Asterisk |
| 22:49.51 | monsterco | braden80 ^^^ |
| 22:51.50 | braden80 | hm i dunno i'm not an expert |
| 22:52.23 | ChannelZ | Last *called* or last *caller* ? |
| 22:53.25 | ChannelZ | and you said something about a fax machine, so you're trying to dial from a fax machine hooked up to the ATA? Is it dialing too fast? |
| 22:55.37 | monsterco | ChannelZ - Last Called Number: |
| 22:55.47 | monsterco | and yes it is connected to a FAX machine which did work before |
| 22:55.53 | monsterco | so I doubt it's dialing really quick now |
| 22:56.13 | monsterco | I am not on site right now - I have to try to find a way to make the Fax send a test one now |
| 22:56.45 | monsterco | is there any way to see more logs on ATA? as to why this is happening? I saw the same 47 in asterisk verbose and most likely is the same in SIP packets |
| 23:00.01 | braden80 | are you manually dialling or speed dialling |
| 23:00.10 | ChannelZ | not that I know of. The ATA is either misreading the DTMF and only picking up a few digits, or the FAX machine is busted |
| 23:00.46 | ChannelZ | unless you had a substitution dialstring in the ATA causing it to only send 47 specifically or something.. |
| 23:02.43 | ChannelZ | brb must poo badly |
| 23:03.07 | monsterco | braden80 - manually dialing |
| 23:03.38 | monsterco | both FAX and ATA were reset at one point - so I am not sure what is the culprit now |
| 23:03.47 | WIMPy | That's called a core dump in here. |
| 23:03.59 | monsterco | I am wondering if there is a way to activate the fax to send me one - I am remote... |
| 23:04.01 | braden80 | lol |
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| 23:17.59 | ChannelZ | whew.. core dump for sure, almost had a buffer overflow |
| 23:18.54 | braden80 | lol |
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