00:33.00 | Samot | drmessano: |
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01:58.07 | budman_mtl | hello ppl |
01:58.39 | budman_mtl | i need some advise on a project creation |
01:59.31 | budman_mtl | need to use asterisk to call ppl from db list and have it run a scripr according to user dtmf entry |
01:59.39 | budman_mtl | *script |
02:00.24 | budman_mtl | any starting point would be apreciated, i did some project involving asterisk, but i'm a n00b. thxs in advance |
02:02.02 | budman_mtl | ps this would call internal extention only, but i dont think it make a diff to asterisk . |
02:05.13 | budman_mtl | plz pm if you have any info, ill be around. Thxs |
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02:36.34 | [TK]D-Fender | ~book |
02:36.34 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
02:36.35 | [TK]D-Fender | ^^^ |
02:36.55 | [TK]D-Fender | For the callouts : AMI Originate, or Call Files. |
02:37.00 | [TK]D-Fender | ~asteriskwiki |
02:37.00 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
02:37.02 | [TK]D-Fender | ^^^^ |
02:37.19 | [TK]D-Fender | budman_mtl, ^ |
02:50.09 | budman_mtl | thxs |
02:50.19 | budman_mtl | will do my homework |
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08:18.41 | Milos | What is this bridge_simple.so which was added/forced in Asterisk 13? |
08:20.19 | Milos | http://infobot.rikers.org/%23asterisk/20140829.html.gz |
08:20.31 | Milos | Apparently "bridging has been split out into modules" |
08:20.41 | Milos | mentioned by file back at the end of 2014 |
08:21.20 | Milos | Can I get some more context on that? If it was split out into modules, I would have expected to still see bridges being joined in the Asterisk 11 logs, but it seems to be new in Asterisk 12+ (xxx joined 'simple_bridge' basic-bridge <537840ed-ef7e-446c-91db-17632e159acc>) |
08:21.30 | Milos | What is simple_bridge and/or basic-bridge exactly? |
08:25.14 | Milos | https://wiki.asterisk.org/wiki/display/AST/Bridges - is this just an internal thing that you've decided to make a public interface for? |
08:25.30 | Milos | "Dial - a bridge is created for the two channels when the outbound channel answers. Both the inbound channel and the outbound channel are placed into the bridge." |
08:25.41 | Milos | Obviously this was possible before Asterisk 12, hence my question. |
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09:18.51 | Samot | Asterisk 12 is EOL |
09:19.38 | Milos | I never used Asterisk 12. |
09:19.48 | Milos | Made the switch from 11 to 13. |
09:22.26 | Samot | What is the issue you are having? |
09:23.12 | Milos | No issue - just wondering what the new bridge stuff is and how it relates to Asterisk 11. It seems in 12+ a bridge is created for each call. Calling works in 11 without these bridges, so what gives? |
09:23.48 | Samot | Well a bridge happens when there are two or more channels that share media.. |
09:24.06 | Samot | So yes, when you make a call from your device to another device, those two channels are bridged together. |
09:25.12 | Milos | I had to add load => bridge_simple.so to make this work, since I only load the modules that I choose. Now, logs mention (xxx joined 'simple_bridge' basic-bridge <537840ed-ef7e-446c-91db-17632e159acc>) which again wasn't seen in 11 and below. |
09:25.27 | Milos | My question is, is this a core feature that was hidden from the end user, now made into a module, or what? |
09:26.43 | Samot | Well bridging happened in Asterisk 11 |
09:26.46 | Samot | And before. |
09:26.51 | Milos | To rephrase my question: In Asterisk 11, calling worked fine without any mention of bridges neither as a module nor in the regular logs. In 12+ this module has to be loaded and the logs talk about creating a bridge for new calls. |
09:27.00 | Milos | So does that mean it was an internal feature that was made public in 12+ ? |
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09:35.53 | Samot | I'm not sure what you mean, but I just found an Asterisk 11 box and bridge_simple.so is a module that needs to be loaded. |
09:38.20 | Samot | In fact, going back to a box running 1.8, bridge_simple.so is a module that needs to be loaded. |
09:38.26 | Samot | So this isn't a new change. |
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20:00.19 | bravvve | hi am having some truble to conect to my server from wan,need some help pleeze |
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20:04.48 | WIMPy | We have a strict policy here. You won't get any answers without a question. |
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20:14.51 | bravvve | WIMPy:some help in testing !!!! |
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20:22.25 | Samot | bravve: Then actually state what you are trying to achieve, what you have actually done and what type of errors are you getting. |
20:23.00 | Samot | Until you actually tell us the issue and what you are doing, we will not...er..cannot help. |
20:29.53 | bravvve | am having asterisk server installed behined DSL modem under a openvz host machine,i can connect from machine in LAN network,but not WAN,i can't figure why |
20:31.30 | bravvve | ip is 41.100.88.172 tcp 5062,account name 2201 password p2201 |
20:32.11 | RS232C | bravvve, have you forwarded the udp port to your internal firewall? |
20:32.20 | RS232C | err server |
20:33.08 | bravvve | tcp,yes |
20:33.09 | RS232C | or are you intentionaly using tcp? |
20:34.48 | igcewieling | udp is not tcp. |
20:34.55 | bravvve | using wireshark,i get resquet from distant machin,and the host is sending reponce,but nothing in the server |
20:35.42 | RS232C | nmap says your port is closed |
20:35.45 | bravvve | yes i know udp,and tcp,am using tcp to folow ACKs |
20:36.24 | bravvve | ops,5060 not 5062 |
20:36.48 | RS232C | ya that shows open |
20:37.46 | RS232C | hmmm I have never used tcp for sip |
20:38.22 | RS232C | wireshark should be able to follow a udp sip conversation... |
20:38.46 | bravvve | am using tcp cause it should give ACKs |
20:38.53 | RS232C | I assume it can follow a tcp one as well |
20:39.05 | bravvve | udp do no |
20:39.40 | RS232C | have you turned on "sip set debug on" in your asterisk console? |
20:41.16 | RS232C | you might also need something like "nat=force_rport,comedia" in your device config in sip.conf |
20:41.35 | RS232C | but I don't know if that applys to tcp |
20:42.32 | bravvve | what nat=force_rport.comedia do? |
20:43.31 | RS232C | nat = force_rport ; Pretend there was an rport parameter even if there wasn't |
20:43.40 | RS232C | nat = comedia ; Send media to the port Asterisk received it from regardless |
20:44.45 | bravvve | rport? |
20:44.46 | RS232C | but I think thats for media once sip is established |
20:44.55 | RS232C | remote port |
20:45.16 | bravvve | yes ,the problem with me that i can't connect |
20:45.44 | bravvve | RS232C:can connect to my server? |
20:46.07 | bravvve | RS232C:from your location???? |
20:46.22 | RS232C | about to try |
20:47.53 | bravvve | ok |
20:48.25 | RS232C | my ciscoSPA525G says registration failed |
20:49.36 | bravvve | how can do virbos mode ? |
20:50.29 | RS232C | in asterisk -r "core set verbose 999" and "core set debug 999" |
20:51.02 | Samot | A core debug isn't needed at this point. |
20:51.19 | Samot | And it will produce a bunch of unneeded output. |
20:51.47 | RS232C | I usually turn it on. but your right |
20:52.07 | Samot | 1) Are all the ports forwarded properly from the router to Asterisk? |
20:52.23 | Samot | 2) Are things like SIP ALG or SPI Firewall disabled in the DSL router? |
20:52.49 | Samot | 3) Are the external ip, local_networks, etc configured properly? |
20:54.41 | Samot | Until all the NAT/network stuff is configured properly in Asterisk, doing a debug for remote connections is kinda pointless. |
20:54.57 | Samot | And in the router. |
20:55.49 | bravvve | hhh,am using GNS to test network stuffs |
20:56.00 | RS232C | bravvve, is your setup like this ... phone <--> firewall <--> internet <--> firewall <--> asterisk ? |
20:56.13 | Samot | I don't know what GNS is. |
20:56.33 | RS232C | cisco router emu |
20:59.32 | bravvve | yes is Cisco router emulator,but i get to configure a router too hhhhhhh,and am using openvz,too many things to configure hhhhhhh |
20:59.57 | bravvve | and at last see what happend with wireshark |
21:05.54 | RS232C | bravvve, I am assuming you are running asterisk on some for of linux? |
21:06.26 | RS232C | form even |
21:06.57 | RS232C | you might check your firewall setting on that server as well |
21:09.37 | bravvve | yes a debian vz template,some cained of virtualization |
21:10.13 | RS232C | so I am working with 2 asterisk servers one is voice mail... but I can't seem to get "mwi => 1234:password@mysipprovider.com/1234" working... I can get voice mail and remove voice mails but MWI is not working. any gotchas I should be aware of? |
21:10.57 | RS232C | also the main server was compiled for imap support but not file storage support |
21:11.28 | RS232C | but not sure if that impacts MWI |
21:13.09 | RS232C | bravvve, you might check is it is blocking non local access to tcp 5060 |
21:13.43 | RS232C | s/is/if/g |
21:14.12 | RS232C | lol |
21:15.19 | RS232C | infobot, you almost got it right! |
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22:29.05 | bravvve | RS232C:hi |
22:29.16 | RS232C | yo |
22:31.12 | bravvve | can help to test with 41.100.40.122:5060 username 2201 password p2201,plz |
22:31.18 | bravvve | can help to test with 41.100.40.122:5062 username 2201 password p2201,plz |
22:31.27 | RS232C | tcp still? |
22:31.27 | bravvve | 5062 not 5060 |
22:31.32 | bravvve | tcp yes |
22:31.37 | RS232C | ya |
22:33.05 | RS232C | loons like its not registring |
22:35.18 | bravvve | and now? |
22:35.33 | RS232C | hold on turning up syslog and rebooting phone |
22:37.42 | RS232C | looks like its not connecting |
22:38.40 | bravvve | ok |
22:40.03 | RS232C | Dec 11 16:42:53 10.42.0.199 [1:0]SIP/TCP:ConnErr:110 |
22:40.03 | RS232C | Dec 11 16:42:53 10.42.0.199 [1:0]SIP/TCP:Connect=-1 |
22:40.03 | RS232C | Dec 11 16:42:53 10.42.0.199 [1]SIP/TCP Backoff 500 ms |
22:40.03 | RS232C | Dec 11 16:42:53 10.42.0.199 [1]SIP:RegFailed;Retry in 30s |
22:40.20 | RS232C | thats the syslog from my phone |
22:41.39 | RS232C | oh I just noticed a diff ip... my bad |
22:42.13 | bravvve | ok,give it again |
22:42.22 | bravvve | or hit it again |
22:42.49 | RS232C | 5060 or 5062? |
22:43.40 | RS232C | 5062 is failing a lot faster now |
22:43.41 | bravvve | 5062 |
22:45.55 | bravvve | can tray again plz |
22:47.24 | RS232C | still fail |
22:47.34 | RS232C | SIP/TCP:ConnErr:111 |
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22:52.21 | bravvve | not getting you at all |
22:53.05 | RS232C | nothing from 216.106.72.202 ? |
22:54.17 | bravvve | no ,and its working very well between 2 routed networks,i'v tested it with gns3 |
22:55.48 | bravvve | now the problem is in the router or ISP is blocking it |
22:55.56 | bravvve | am getting 211.103.199.98 |
22:56.07 | bravvve | am getting 211.103.199.98,but http not 5062 |
22:56.56 | RS232C | this feels like a firewall issue to me |
22:57.22 | bravvve | port is open ,have tested it |
22:59.02 | RS232C | ... what country are you in? |
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23:02.22 | RS232C | that ip says Algeria... mabe a government level block? |
23:03.56 | RS232C | I know some countries block voip |
23:05.26 | RS232C | if thats the case you may have to tunnel though a VPN and hope that is not blocked |
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