| 00:26.51 | RadicalDev | Is there a way I can get a list of all the active channels in asterisk besides core show channels? |
| 00:27.22 | [TK]D-Fender | AMI |
| 00:30.05 | RadicalDev | I don't see anything about that in the AMI docs. |
| 00:32.57 | RadicalDev | I've been doing ami.command("core show channels verbose"), but I've run into an issue where the channel name is longer than what is reported with that command. |
| 00:33.36 | RadicalDev | e.g., core show channels verbose lists SIP/outbound-0000032, but the actual channel ID is SIP/outbound-00000327 |
| 00:35.35 | WIMPy | concise might work better than verbose, but you should take a looka the Status command. |
| 00:36.20 | [TK]D-Fender | Correct |
| 00:36.21 | WIMPy | Apart from that you can just keep listening and will always know what's going on. |
| 00:36.29 | [TK]D-Fender | concise lists the full with a delimiter |
| 00:36.38 | [TK]D-Fender | and AMI has direct functions. |
| 00:36.47 | [TK]D-Fender | You were just using AMI ... to call a CLI command |
| 00:36.56 | [TK]D-Fender | Which is not why I suggested AMI |
| 00:37.02 | RadicalDev | I was in a hurry some years ago when I wrote that =) |
| 00:37.24 | [TK]D-Fender | using AMI to call the CLI command is not an "alternative" or "better" .... than just calling the CLI command you asked for an alternative to... |
| 00:37.50 | RadicalDev | Hence my first response: RadicalDev> I don't see anything about that in the AMI docs. |
| 00:37.54 | WIMPy | Well, it does allow for better permission management. |
| 00:38.02 | WIMPy | But yeah, it's a pretty bad idea. |
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| 00:47.40 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_CoreShowChannels |
| 00:47.41 | [TK]D-Fender | ^^^^^ |
| 00:47.42 | rmudgett | RadicalDev: Look at "manager show command CoreShowChannels" |
| 00:48.01 | [TK]D-Fender | Which gives you : https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_Status |
| 00:48.03 | [TK]D-Fender | ^^^^^ |
| 00:49.54 | RadicalDev | this box is asterisk 1.8 =/ concise works though. |
| 00:52.37 | rmudgett | RadicalDev: That AMI action exists in v1.8 |
| 00:52.46 | [TK]D-Fender | 1.8 is no longer even supported |
| 00:53.14 | [TK]D-Fender | rmudgett, Indeed |
| 00:53.44 | RadicalDev | rmudgett: I'm trying to figure out how to use it. pyst2 has that command, but it is just returning Success, which is unhelpful. |
| 00:54.05 | [TK]D-Fender | Did you read the second link? |
| 00:54.47 | [TK]D-Fender | First triggers the transmission of those following events |
| 00:55.08 | RadicalDev | Yep. I figured that out right before you said it =) |
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| 01:07.50 | EllenorL | hugs [TK]D-Fender for no apparent reason |
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| 02:17.01 | socomm | I've compile pjsip into asterisk but when I try to connect an sip client to asterisk it doesn't take. |
| 02:17.14 | socomm | nmap shows 5060 as closed, but netstat shows it as listening |
| 02:17.27 | socomm | when I telnet to 5060 i get connection refused - locally |
| 02:18.11 | WIMPy | SIP usually uses UDP. |
| 02:18.22 | socomm | sorry, mispoke with listening state |
| 02:18.47 | socomm | but netstat does see it as bound to the LAN address |
| 02:19.04 | socomm | I've already flushed firewall rules for testing purposes |
| 02:19.39 | socomm | I've even have debugging on, but the connection doesn't seem to hit asterisk |
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| 02:24.32 | [TK]D-Fender | Telnet is not going to work unless you specifically configured * to use TCP and not the default of UDP |
| 02:26.24 | socomm | [TK]D-Fender: right you are |
| 02:59.03 | wyoung | EllenorL: You don't need a reason |
| 02:59.22 | EllenorL | wyoung, i do if he thinks i do |
| 02:59.36 | wyoung | [TK]D-Fender: tcpdump would help |
| 02:59.42 | wyoung | or netcat |
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| 03:36.45 | jfindley | or ssh, tcpdump, and wireshark... ssh user@host "tcpdump -U -s0 -w - not port 22" | wireshark -k -i - |
| 03:39.45 | LiuYan | asterisk-ldap is used for realtime configuration only, not for providing LDAP function to fetch information from LDAP server, right? |
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| 04:39.16 | Samot | So I have this in my dialplan: exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@${provider},43,L(20000)) limits the call to 2 minutes but I want to set the variable LIMIT_PLAYAUDIO_CALLER to no. I'm not sure how to set that option in the Dial() string. Is it in the L(20000) option? Anyone know? |
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| 05:39.39 | [TK]D-Fender | Samot, You don't set it in Dial. You set it BEFORE Dial. |
| 05:40.42 | Samot | OK. Thanks. |
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| 05:58.42 | EllenorL | Set it in hell. |
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| 06:25.33 | jfindley | Is there a way to exit from an AGI script and go to a different context,exten,priority? |
| 06:27.02 | [TK]D-Fender | Yes, there is an AGI command expressly for that |
| 06:30.58 | jfindley | gosub? |
| 06:39.12 | [TK]D-Fender | No, that implies going back |
| 06:39.22 | [TK]D-Fender | you simply set the context, exten & priority |
| 06:39.28 | [TK]D-Fender | and end it |
| 06:39.34 | [TK]D-Fender | heads off to bed |
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| 07:16.29 | utrack | Hey, could anyone help me with AMI? |
| 07:16.36 | utrack | Simple question |
| 07:16.41 | utrack | I'm using these settings: http://pastebin.com/RDmYGNUw |
| 07:17.03 | utrack | However when I telnet to AMI and login I can't see any userevents fired after the call |
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| 07:29.49 | utrack | Even after adding the g option to the Dial() :( |
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| 07:36.18 | wdoekes | utrack: that's not a valid dialplan |
| 07:37.10 | wdoekes | $ asterisklint dialplan-check utrack.conf |
| 07:37.11 | wdoekes | utrack.conf:3 W_DP_PRIO_BADORDER: bad priority order for pattern 's' and prio <unset> |
| 07:37.35 | wdoekes | you should replace "exten=>s," with "same=>" |
| 07:37.56 | wdoekes | or "exten=>_XXXXXXXXXXX," |
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| 07:52.41 | utrack | wdoekes: thanks! I'll try |
| 07:53.58 | utrack | wdoekes: works, thank you :) |
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| 08:32.06 | utrack | Is there any way to extract the answer's status and who hung up the call first? ${HANGUPCAUSE} is 0 (undefined) if the peer didn't answer |
| 08:36.06 | utrack | nvm, found it |
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| 08:53.07 | jfindley | utrack: What did you find? |
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| 09:48.21 | utrack | jfindley: ${DIALSTATUS} returns the answer's status more reliably than ${HANGUPCAUSE} :) |
| 09:48.48 | jfindley | Yeah, but no indication of who hung up |
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| 12:08.40 | utrack | jfindley: oh, right :( that I still don't know |
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| 14:47.37 | deepend | <PROTECTED> |
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| 15:06.08 | dan_j | Hi. If a caller joins the queue but then hangs up before answering, in the 'h' extension, what variable can I use to determine that the call was never answered by an agent? |
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| 16:19.53 | exuberocity | Morning -- quick question about a PRI. Anyone able to answer a quick question? I have a program called GFI faxmaker that has virtual "trunks" that go over a PRI. At a certain point the trunk is "requesting the PRI go into maintenance mode" per our carrier. Is this something that can happen? Anyone have any idea? |
| 16:20.20 | exuberocity | Just dead air when you call the number when it happens and when you try to fax out from the program it's like it's looping back. |
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| 16:26.19 | WIMPy | What does maintenance mode mean? Is that just a phrase for "down"? |
| 16:27.31 | WIMPy | But no, surely not normal. |
| 16:29.30 | exuberocity | Yeah, "down" is what they are and the carrier calls it maintenance. |
| 16:29.52 | exuberocity | I would think the carrier's equipemnt would be able to ignore maintenance signals or specified requests as needed. |
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| 18:46.39 | Bhakimi | hi guys, i have two servers on a local network both running asterisk 11.22 and i've ben having issues with iax trunks giving me all kinds of issues randomly. i jsut discovered that there is loss and jiter between the boxes when there is a iax2 trunk. i ran iax2 show netstats to get this information, any idea how i can trace this issue? there is absolutely no packet loss or latency between the two servers |
| 18:47.24 | Bhakimi | the message that floods asterisk and causes it to stall for a few minutes is "Exceptionally long voice queue length queuing to" |
| 18:48.55 | Bhakimi | http://pastebin.com/kiu6UqqD is a link to the netstats output |
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| 19:14.14 | *** join/#asterisk acro458 (0cab5efe@gateway/web/freenode/ip.12.171.94.254) |
| 19:17.09 | acro458 | Hello, I haven't used asterisk yet. I want to know if I can use it to develop an application. I want someone to be able to call a phone number, the application will pick up, and get "Thank you for calling xxxx press 1 for xxxxx, press 2 for yyyy, etc" and respond based on key pressed. |
| 19:17.21 | acro458 | Is this something I can use asterisk for? |
| 19:17.32 | acro458 | (or is there a better/easier solution) |
| 19:19.09 | acro458 | I would like to run this application from a computer with only an ethernet cable connected (no POTS line) |
| 19:21.16 | acro458 | I have seen this page: http://www.asterisk.org/get-started/applications/ivr |
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| 19:23.17 | acro458 | my main question is how do I set it so when someone calls my AT&T phone number, the asterisk application picks up (and plays the voice message) |
| 19:26.34 | [TK]D-Fender | Yes |
| 19:26.49 | [TK]D-Fender | * is a good platform for this |
| 19:27.14 | [TK]D-Fender | Call comes in from "wherever" and you can have it go through whatever hoops you want it to |
| 19:29.02 | acro458 | Ok |
| 19:29.09 | Bhakimi | acro458 you need a fxo card and place that in your server |
| 19:29.19 | Bhakimi | you can use sangoma for that, they make usb boxes for that |
| 19:29.42 | Bhakimi | and you wanpipe with asterisk to send that call into asterisk, then direct that call to a ivr |
| 19:30.08 | Bhakimi | setup a freepbx box, they have nice UIs that allow you to configure all of that |
| 19:31.06 | acro458 | Ok, so I can't just use a single ethernet port? (when I use google voice from computer to computer, I do not need any special hardware) |
| 19:31.21 | acro458 | I assume Google uses the special hardware internally? |
| 19:31.31 | Bhakimi | if its a google voice number you can probably use gtalk module in asterisk |
| 19:31.37 | Bhakimi | i thinkt hey have it in asterisk 11 or 13 |
| 19:32.00 | Bhakimi | and then forwoard ur at&t number to the google number |
| 19:32.43 | acro458 | It is not, it is an AT&T pots line. |
| 19:33.02 | acro458 | I am just hoping for a solution without any special hardware |
| 19:33.51 | acro458 | It needs to be able to support at least 50 concurrent incoming calls (no outgoing) |
| 19:34.27 | Bhakimi | you can send that number to a google voice number and have asterisk register to the google voice number |
| 19:34.36 | Bhakimi | or get a cheap inbound sip did |
| 19:34.46 | Bhakimi | purchase a voip number |
| 19:34.51 | Bhakimi | register that number on ur pbx |
| 19:34.58 | Bhakimi | and forwoard ur land line to that number |
| 19:36.27 | acro458 | hehe, you lost me. :) |
| 19:36.42 | Bhakimi | ok so purchase a voip number |
| 19:36.43 | acro458 | but I copied what you said because after research it may make sense |
| 19:36.46 | Bhakimi | from a voip carrier |
| 19:36.57 | Bhakimi | say twillio for example |
| 19:37.08 | acro458 | I used them once years ago |
| 19:37.12 | Bhakimi | and you register that number on your asterisk box |
| 19:37.40 | Bhakimi | once you register to twillio, anytime you call that number twillio will send that call to your server |
| 19:37.52 | Bhakimi | this way u have the ability to send calls to your server |
| 19:38.00 | Bhakimi | then point that twillio number to the ivr |
| 19:38.17 | Bhakimi | once thats done point your at&t number to the twillio number |
| 19:38.27 | acro458 | ok that makes sense |
| 19:38.29 | Bhakimi | im sure at&t can forward calls |
| 19:38.36 | acro458 | yah, we can change forwards |
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| 19:39.31 | acro458 | But I have to use a service such as twillio, correct? |
| 19:40.06 | acro458 | (in order to not need special hardware) |
| 19:40.28 | Bhakimi | <--- needs help figuring out why i have jitter and packet loss between 2 IAX trunks, both servers are on a local network and there is no latency OR packet loss in between them. here is a link tot he iax2 show netstats results : http://pastebin.com/kiu6UqqD |
| 19:40.55 | Bhakimi | acro458 you may be able to use google voice, i think asterisk has a plugin that allowes you to register a google voice numnber to it |
| 19:40.59 | Bhakimi | its asterisk 11 or 13 |
| 19:41.54 | acro458 | ok, Google voice is at least good for testing. |
| 19:42.02 | acro458 | I will give it a try |
| 19:42.03 | acro458 | Thanks! |
| 19:42.06 | file | Google Voice may or may not work, and they don't like you using it for multiple calls... it's really for average user usage |
| 19:42.06 | Bhakimi | np =) |
| 19:42.44 | Bhakimi | @file you have any experience dealing with iax trunks ? |
| 19:43.07 | file | long ago, I don't touch it these days |
| 19:43.48 | Bhakimi | i upgraded our farm of servers to asterisk 11 and im getting random iax2 issues and i jsut found out every iax trunk has exacly a lost of 40 packets which is strnage |
| 19:48.53 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:04.29 | *** join/#asterisk BlackMaria (~BlackMari@dsl.198.58.150.55.ebox.ca) |
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| 20:40.20 | lorsungcu | I've got a SIP provider that requires I have the private IP of the PBX in the contact header of invites sent to a publicly routed address. Is there any way to do that, outside of adding their proxy address to localnets? |
| 20:41.17 | lorsungcu | using chan_sip |
| 20:41.34 | lorsungcu | asterisk 13.5 |
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| 22:22.52 | lanning | when did vm-*.ulaw disapper from asterisk-sounds-core-en? |
| 22:23.00 | *** join/#asterisk EllenorL (~3113102@unaffiliated/ellenor) |
| 22:27.40 | lanning | ok, so it is in the tar.gz but not in asterisk-sounds-core-en-1.4.25-1.el6.noarch.rpm |
| 22:58.02 | *** part/#asterisk kharwell (kharwell@nat/digium/x-muwmlfigdvxckrye) |
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