| 00:16.27 | muanang | dahdi_config module fixed my echo problem! thanks for the help! |
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| 02:26.52 | phix | muanang: you were having dahdi issues? |
| 02:26.56 | phix | I am having dahdi issues! |
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| 04:41.13 | ChannelZ | try a therapist |
| 04:41.19 | ChannelZ | AHhhAAHAA |
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| 10:30.17 | babak | Hi, anyone knows TeleYapper ? |
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| 11:49.37 | ldc | hi! http://pastebin.com/HLiqQdsz anyone else getting this? |
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| 11:57.05 | imihaylov | Hello all! I want to ask is it possible to call subrouting in the dial command? |
| 11:57.17 | imihaylov | something like a Ð(macro) |
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| 12:22.15 | anonymouz666 | hello |
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| 12:50.09 | anonymouz666 | Fender stratocaster is in da house |
| 12:54.04 | anonymouz666 | SIP offers a playlist to [TK]D-Fender |
| 12:55.34 | [TK]D-Fender | anonymouz666: My nick had nothing to do with music, and I've only owned about 2 Fender products ever and only 1 item recently. |
| 12:56.18 | anonymouz666 | got a 488 Not Acceptable here |
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| 12:58.57 | [TK]D-Fender | Make a better offer |
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| 15:56.19 | xpheres | hello, I have problems receiving calls from a did number from buyddinumber.com |
| 15:56.28 | xpheres | I have this message in the log: Warning: 399 Bad MWI NOTIFY |
| 15:56.35 | xpheres | could someone help me please? |
| 15:57.16 | WIMPy | That is not related to a call.Well, nNot a current one. |
| 15:57.29 | xpheres | mm |
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| 16:21.14 | structur | greetings all |
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| 17:00.35 | marceloamorim | guys, there is any issue related between "can't send 10 type frames with SIP write" and vmware machine? |
| 17:04.47 | xpheres | anyone has problems with buyddinumber ? |
| 17:04.55 | xpheres | I have this message in the log: Warning: 399 Bad MWI NOTIFY |
| 17:05.35 | anonymouz666 | xpheres: MWI means message waiting indication |
| 17:05.45 | xpheres | I have a cisco 7960 |
| 17:05.49 | anonymouz666 | probably some client is sending a MWI that asterisk can't parse |
| 17:05.51 | xpheres | I used it before with a did number |
| 17:05.56 | xpheres | mm |
| 17:05.58 | xpheres | why? |
| 17:06.13 | anonymouz666 | xpheres: you have to discover why. |
| 17:06.22 | xpheres | I have nat no in the extension and sip configuration |
| 17:06.29 | anonymouz666 | marceloamorim: only in vmware? |
| 17:06.41 | xpheres | I followed this to configure the trunk : https://www.buyddinumber.com/DDI-Numbers-freepbx-configuration.html |
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| 17:11.33 | marceloamorim | anonymouz666: actually seems like, I'm transfer all my codes to this vmware, when I was testing it seems fine, but when I actually migrated my main asterisk, the audio was a mess and I was getting this "can't send 10 type frames with SIP write" |
| 17:11.37 | marceloamorim | so I moved back |
| 17:12.13 | xpheres | please |
| 17:12.23 | xpheres | just in case anyone has the experience to help me understand |
| 17:12.44 | xpheres | why if I cal my did number, I receive the call in my telephone after the caller has hanged up? |
| 17:13.45 | WIMPy | You don't. |
| 17:13.57 | WIMPy | You are informed that the caller left a voice mail. |
| 17:14.27 | xpheres | no |
| 17:14.40 | xpheres | the receiving telephone rings |
| 17:14.51 | xpheres | but the caller has already hanged up |
| 17:15.36 | xpheres | maybe if I post the log someone could understand the reason? |
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| 17:15.54 | WIMPy | Maybe you should the whole situation.escribe |
| 17:16.07 | xpheres | what |
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| 17:28.54 | xpheres | hi |
| 17:29.16 | xpheres | could anyone understand with this why I receive the call after the caller has hanged up? |
| 17:29.17 | xpheres | http://pastebin.com/MmQ9Wun2 |
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| 17:56.45 | babak | Hi, anyone knows Teleyapper ? |
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| 18:00.59 | [TK]D-Fender | ~polls |
| 18:00.59 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 18:02.59 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.68.40) |
| 18:07.32 | babak | has any one source code of TeleYapper PhoneBook ? http://pbxinaflash.com/community/index.php?threads/teleyapper-phonebook-v5-01-beta.8492/ |
| 18:22.29 | *** part/#asterisk xpheres (~bodhi64@dslb-088-073-046-091.088.073.pools.vodafone-ip.de) |
| 18:31.39 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
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| 19:07.01 | *** join/#asterisk structur (~user@unaffiliated/structur) |
| 19:08.39 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:18.38 | *** join/#asterisk t4nk579 (809fffe2@gateway/web/freenode/ip.128.159.255.226) |
| 19:21.34 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 19:22.04 | t4nk579 | Hello all. I have been using Asterisk for a few months with WebRTC but about a month ago I was no longer receiving audio. I can still see the call being made and RTP debug shows packets flowing, but I don't get any audio. I set up Freeswitch to see if it would work and indeed it does, but I would prefer to get Asterisk working again. |
| 19:27.37 | *** join/#asterisk roop-jr (~roop@ip-4a0f.proline.net.ua) |
| 19:31.09 | roop-jr | Hello. In Asterisk 13 was modified ANSWEREDTIME calculation. In my usage problem occurs when I answered the call for use playback. Detailed: https://issues.asterisk.org/jira/browse/ASTERISK-24943 |
| 19:31.29 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 19:31.55 | roop-jr | Can someone help me? How I can circumvent this innovation? |
| 19:35.12 | marceloamorim | I paste that problem I said early http://pastebin.com/4CwKBqQn. I don't know what kind of delay is going on, could you see the problem with this pastebin? |
| 19:36.39 | *** join/#asterisk Nny (ae6bd1db@gateway/web/cgi-irc/kiwiirc.com/ip.174.107.209.219) |
| 19:37.30 | [TK]D-Fender | marceloamorim: Next time don't include 2500 lines of garbage we don't need at the front. |
| 19:37.36 | Nny | Quick question. I have a Goto statement. The context it is sent to has a pattern match. Do I need to define the extension specifically that the goto points to or will the pattern match work? |
| 19:37.51 | [TK]D-Fender | marceloamorim: You also haven't explained WHERE this delay is happening. You've given no detail with your description |
| 19:38.03 | [TK]D-Fender | Nny: Should hit the pattern |
| 19:39.01 | Nny | Thanks |
| 19:39.06 | *** join/#asterisk sommarnatt (~sommarnat@c83-248-69-249.bredband.comhem.se) |
| 19:39.48 | marceloamorim | the delay is going on on audio, I use application Playback but the audio seems like robot and sometimes I can't listen. and sorry about my full.log, I forget to remove the asterisk's start. |
| 19:40.43 | marceloamorim | I called from 401 extension to 01@marceloteste |
| 19:40.57 | [TK]D-Fender | how LONG is the delay? |
| 19:41.25 | [TK]D-Fender | Do go thinking we can see it actually happening as to when you actually hear it start (if at all) |
| 19:41.46 | marceloamorim | sometimes is the whole application, I can't even hear anything, but sometimes is like robot so I listen all audio but with robot kind |
| 19:42.34 | marceloamorim | whole application playback* |
| 19:42.44 | [TK]D-Fender | [Apr 6 16:23:46] DEBUG[13249][C-00000000] format_wav.c: Skipping unknown block 'LIST' |
| 19:42.45 | [TK]D-Fender | ^^^ |
| 19:42.54 | [TK]D-Fender | So far your files don't look like they are in good condition. |
| 19:43.59 | marceloamorim | what does it means? |
| 19:44.59 | marceloamorim | if I use a sip client in the same network, this problem didn't happen |
| 19:45.07 | [TK]D-Fender | It means your files are NOT in good condition. |
| 19:48.35 | marceloamorim | I'll check the wav parameters but why when I use other sipphone nothing wrong ( apparently ) happens ? |
| 19:55.14 | *** join/#asterisk bobbymc (~bobbymc45@208.78.137.2) |
| 19:55.50 | *** join/#asterisk justsomedood (~warren@mail.serverplus.com) |
| 20:03.13 | *** join/#asterisk joecool (~joecool@no-sources/joecool) |
| 20:15.04 | marceloamorim | weird =( |
| 20:16.39 | *** join/#asterisk mjordan (mjordan@nat/digium/x-mkrmawuyxkmxlmxp) |
| 20:16.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:17.01 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
| 20:34.18 | *** join/#asterisk jvwjgames (~jvwjgames@c-174-52-179-61.hsd1.ut.comcast.net) |
| 20:34.25 | jvwjgames | Hi |
| 20:34.58 | jvwjgames | I am having trouble making outbound calls on my pbx i am using ipcomms as the sip provider |
| 20:35.37 | jvwjgames | and the trunk is registered and everything but my box is not responding to the 401 requests from the provider |
| 20:40.32 | eschmidbauer | 407 requests? |
| 20:40.59 | *** part/#asterisk justsomedood (~warren@mail.serverplus.com) |
| 20:41.10 | marceloamorim | guys, when tk said to me was the file, if I sox -V this file can I see if there any corrupted? http://pastebin.com/ARxG3zsD |
| 20:41.37 | jvwjgames | no 401 |
| 20:42.02 | eschmidbauer | that sounds like a problem with your sip provider |
| 20:42.12 | eschmidbauer | they are telling you the call is unauthorized |
| 20:43.19 | jvwjgames | this is what they said |
| 20:44.32 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
| 20:45.08 | jvwjgames | We are getting the call but your side is not responding correctly to the 401 back from us, when we get your call invite, our getway sends you a 401 unauthorized after your first invite attempt, so your side will resend the call invite with authorization. but it's sending the registration packet instead of the 2nd leg attempt invite. |
| 20:45.08 | jvwjgames | <PROTECTED> |
| 20:45.08 | jvwjgames | Not sure if this is a bug in your FreePBX/Asterisk version. you may need to look at the Asterisk CLI to see what your box is doing when it gets our 401 message. |
| 20:46.40 | eschmidbauer | they should be sending a 407 not a 401 |
| 20:46.50 | eschmidbauer | Who is your SIP provider |
| 20:47.07 | jvwjgames | IPCOMMS |
| 20:47.13 | jvwjgames | ipcomms.net |
| 20:49.37 | eschmidbauer | I may be wrong, but I don't think Asterisk will send the INVITE with authorization in response to a 401 |
| 20:49.41 | K0HAX | A 401 should get a response of the same invite packet, but with a response to the challenge included in their 401 |
| 20:49.54 | eschmidbauer | it absolutely will in response to a 407 |
| 20:50.18 | K0HAX | looks at a packet capture quick |
| 20:52.10 | jvwjgames | hmm |
| 20:52.17 | jvwjgames | so what do i do |
| 20:52.25 | jvwjgames | i really want outbound calls to work |
| 20:52.26 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:52.30 | K0HAX | Yeah, my Polycom phone responds to a 401 with the response to a challenge, so Asterisk should too |
| 20:52.31 | jvwjgames | inbound calls work just fine |
| 20:53.39 | K0HAX | Do a packet capture |
| 20:53.53 | jvwjgames | ok |
| 20:54.16 | K0HAX | sngrep is useful for seeing issues with SIP :) |
| 20:54.41 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
| 20:54.53 | jvwjgames | ok |
| 20:55.17 | eschmidbauer | <PROTECTED> |
| 20:56.12 | eschmidbauer | sipgrep is pretty cool |
| 20:56.17 | jvwjgames | ok |
| 20:56.45 | K0HAX | Asterisk sends my phones 401 all the time when they try to register, my SIP provider sends 401 when trying to INVITE too |
| 20:56.54 | K0HAX | it *should* work |
| 20:57.06 | K0HAX | there might be NAT breaking stuff |
| 20:57.08 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:57.22 | eschmidbauer | true -- get the sip capture |
| 20:59.13 | jvwjgames | i have the sip capture |
| 20:59.33 | eschmidbauer | let's see it |
| 20:59.47 | jvwjgames | it is a wireshark file anyone want it |
| 20:59.55 | jvwjgames | how do i send it |
| 21:00.05 | eschmidbauer | you can post those online |
| 21:00.08 | robmal | ~pb |
| 21:00.09 | 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. |
| 21:02.33 | jvwjgames | it is uploading |
| 21:03.56 | jvwjgames | http://expirebox.com/download/5fad9bdb1ca7be56d2684f61d797d114.html |
| 21:09.09 | jvwjgames | well? |
| 21:16.10 | jvwjgames | did anyone find out what is wrong |
| 21:17.36 | K0HAX | I think you have a NAT problem, you never get a response to your second INVITE |
| 21:17.42 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:17.58 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 21:18.24 | K0HAX | you receive a 401, send an ACK, and then send a bunch of INVITEs with authentication in them, but never receive any packets after the 401. Every time I see that I think NAT |
| 21:18.59 | jvwjgames | hmmm |
| 21:19.00 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:19.28 | jvwjgames | I tried registering my phone on my home network with ipcomms and it works |
| 21:19.34 | K0HAX | if your provider has a kickass support desk they should be able to see a similar packet capture at their end, and they probably don't see the ACK or the second INVITE from you, that or they do and they send something back to you, but you never get it |
| 21:19.48 | jvwjgames | but if i use the pbx witch is on the same network it doesn't |
| 21:21.49 | K0HAX | check that you have nat=no in your sip.conf for the registration to your SIP provider |
| 21:22.03 | K0HAX | yes, it's counterintuitive, but it might fix it |
| 21:22.13 | K0HAX | it fixes mine, but my router has a good SIP ALG in it too |
| 21:22.16 | K0HAX | (most don't) |
| 21:28.35 | jvwjgames | my sip provider recives the registration and also the sip provider says that the nat needs to be set to yes |
| 21:28.40 | jvwjgames | not no |
| 21:28.51 | K0HAX | did you try setting it to no? |
| 21:29.14 | K0HAX | it's easy to change back. :) |
| 21:33.40 | jvwjgames | yes i tired that but still nothing |
| 21:35.19 | K0HAX | hmm |
| 21:35.29 | K0HAX | do a packet capture on the other side of whatever is doing NAT on your network |
| 21:36.11 | jvwjgames | so on the router do a capture of the internet interface |
| 21:37.45 | K0HAX | yeah |
| 21:39.05 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-lbvmptasoqrwtyuj) |
| 21:39.32 | jvwjgames | ok |
| 21:40.07 | K0HAX | btw, I work at a SIP provider. I hate NAT with a burning passion |
| 21:43.23 | jvwjgames | ok |
| 21:43.26 | jvwjgames | i have it |
| 21:43.58 | jvwjgames | and yes this is a INTERNET SIDE ONLY capture and should ONLY HAVE PUBLIC ROUTABLE IPS BUT |
| 21:44.03 | jvwjgames | take a look |
| 21:44.47 | Nugget | NAT blows goats (I have proof) |
| 21:45.02 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 21:45.20 | jvwjgames | http://expirebox.com/download/d4a4c63dfd61a421ae15deeb75d5cb58.html |
| 21:48.59 | *** part/#asterisk mjordan (mjordan@nat/digium/x-mkrmawuyxkmxlmxp) |
| 21:50.54 | K0HAX | lol |
| 21:50.56 | K0HAX | hmm |
| 21:51.08 | K0HAX | does your router have SIP-ALG enabled? If so, try turning it off |
| 21:51.22 | jvwjgames | no pfsense comes with no sip alg |
| 21:51.48 | K0HAX | set externip=<whatever> |
| 21:51.51 | K0HAX | in sip.conf |
| 21:53.14 | jvwjgames | yup it is set |
| 21:54.16 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
| 21:56.06 | K0HAX | try insecure=port,invite in your trunk registration |
| 21:56.34 | K0HAX | if that doesn't work, pfSense is doing something stupid to your SIP packets |
| 21:56.50 | K0HAX | it's not natting them |
| 21:57.34 | K0HAX | in fact.. that's pretty much what I'm going with. pfSense is doing something stupid to your SIP packets. :) |
| 21:59.52 | K0HAX | I wonder if the packet fragmentation has something to do with it... |
| 22:00.10 | K0HAX | it shouldn't.. but sometimes provider SBCs are dumb |
| 22:00.56 | jvwjgames | it is set to that |
| 22:01.26 | K0HAX | jvwjgames: disallow=all, allow=g722,ulaw,alaw |
| 22:01.41 | jvwjgames | set that where |
| 22:01.44 | K0HAX | that should save you some bits in the INVITE and maybe bring you under the 1500 byte MTU |
| 22:01.50 | K0HAX | set that in the trunk registration |
| 22:02.45 | K0HAX | if you try dialing international numbers it might not work because of MTU issues btw. ;) (Isn't SIP awesome!?) |
| 22:03.38 | K0HAX | also, the comma should be a newline. :) |
| 22:03.44 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-lxpkprdogojzlvmm) |
| 22:04.08 | jvwjgames | :) |
| 22:04.12 | jvwjgames | :D |
| 22:04.16 | jvwjgames | works yay |
| 22:04.21 | jvwjgames | thanks so much |
| 22:05.09 | K0HAX | woo! :D |
| 22:05.11 | K0HAX | no problem. :) |
| 22:05.24 | *** join/#asterisk miltux (~miltux@d51A4B2AE.access.telenet.be) |
| 22:06.09 | K0HAX | http://lifeoftechsupport.tumblr.com/post/101367788574/the-mtu-problem |
| 22:06.33 | jvwjgames | i will let ipcomms know to updat there tuts and documentation |
| 22:06.39 | K0HAX | :D |
| 22:06.49 | jvwjgames | cause there documentation states only need ulaw |
| 22:07.19 | *** join/#asterisk zem (~krikkit@dh207-60-89.xnet.hr) |
| 22:07.33 | K0HAX | ah, yes. If you used allow=ulaw only, it should've worked too, but you were allowing everything. If you expand the message body of the INVITE and look at the Media Attributes, you had every codec in there. :) |
| 22:07.51 | K0HAX | the allow= line I gave you lets you do HD voice if your provider ever supports it (mine does) |
| 22:08.04 | K0HAX | alright, brb, going home |
| 22:10.16 | jvwjgames | thanks again |
| 22:23.06 | *** join/#asterisk cyford (~cyford@c-73-207-183-115.hsd1.ga.comcast.net) |
| 22:32.39 | *** join/#asterisk justsomedood (~warren@mail.serverplus.com) |
| 22:33.13 | justsomedood | our asterisk voicemail doesn't play the time of the recording, or when pressing 3-3 for the envelope. Does anyone know why that would happen? |
| 22:33.14 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
| 22:33.26 | justsomedood | Looking through the config it *seems* like it should |
| 22:43.38 | *** join/#asterisk resno (~^_^@unaffiliated/resno) |
| 22:46.04 | resno | when i make calls through my sip trunk the pots person gets lots of breakups. any ideas on what to look at? |
| 22:48.21 | ChannelZ | your upload bandwidth and/or the stability/latency between you and your ITSP |
| 22:49.19 | resno | so the other thing is, if i from the outside make a call in over sip, it sounds okay. |
| 22:49.21 | ChannelZ | Traffic shaping to prioritize the outgoing rtp packets is important if the connection is used for other things, and that those packets make it across the network to your ITSP in a reasonable and consistent amount of time is important |
| 22:49.49 | *** join/#asterisk CeBe (~CeBe@xd9bebb8b.dyn.telefonica.de) |
| 22:50.21 | resno | so, qos is one option i have tried changing |
| 22:50.31 | ChannelZ | Does your ITSP have different routes for incoming and outgoing? (IE do you have two different peers with different IPs/hostnames?) It could be a problem with one of their services |
| 22:50.57 | resno | i dont know, i currently use flowroute |
| 22:51.42 | ChannelZ | well you should know -- do you only have a single peer defined for them in your sip.conf (or pjsip, whatever the case may be)? |
| 22:52.05 | resno | i have a single peer |
| 22:53.47 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 22:54.11 | ChannelZ | OK. Well if you call out to some number and your audio breaks up to them, but that same number calls in to you and your audio doesn't break up to them, there still could be some other routing going on behind flowroute that is broken in some way. |
| 22:54.44 | ChannelZ | If this is consistent, I'd start by checking with them. |
| 22:56.00 | resno | ok |
| 22:57.00 | robl^ | odd question -- but is anyone aware of any SIP ATA (FXS) gateways that can support OLD analog phones with rotary dial? |
| 22:57.13 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 22:58.20 | ChannelZ | hmm |
| 22:59.09 | robl^ | have a few antiques that I want to provide dialtone to at home ;-) |
| 23:00.07 | K0HAX | robl^: I recommend a Cisco IAD2431-8FXS |
| 23:00.18 | K0HAX | It's a tad hard to configure, but they run my rotary phones great |
| 23:00.30 | K0HAX | also, they're relatively cheap on eBay |
| 23:01.07 | robl^ | K0HAX: ohh?? hmm... I'll look into them. thanks for the pointer |
| 23:01.12 | K0HAX | :) |
| 23:03.23 | *** part/#asterisk kharwell (kharwell@nat/digium/x-nrpfjrrossftyqxi) |
| 23:06.19 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
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| 23:29.03 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
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| 23:37.15 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 23:42.26 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.77.202) |
| 23:43.16 | ruben23 | hi guys how od i create a failover dialplan for my asterisk when one VoIP carrier is down it will transfer to other Voip carrier which i setup, i got two setup somehow on my asterisk |
| 23:43.20 | *** join/#asterisk Zer0legend (~ZerOlegen@184-158-70-162.dyn.centurytel.net) |
| 23:43.20 | ruben23 | any idea guys |
| 23:43.34 | K0HAX | for outbound? |
| 23:43.48 | K0HAX | just add a second line under your first Dial() function |
| 23:45.15 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 23:46.45 | K0HAX | inbound requires black magic. :) |
| 23:46.58 | ruben23 | yes for outbound dialing |
| 23:47.13 | ruben23 | can you set an example somehow |
| 23:47.59 | *** join/#asterisk frek818 (~frek818@172.56.17.165) |
| 23:50.29 | ruben23 | K0HAX: ..? |
| 23:52.24 | K0HAX | ok, I had to use a macro.. |
| 23:52.26 | K0HAX | but here |
| 23:52.28 | K0HAX | http://paste.k0hax.com/index.php?id=55231c3044ceb |
| 23:52.30 | K0HAX | :P |
| 23:53.04 | K0HAX | if you use FreePBX there's probably a built-in trunk order you can use on your outbound routes.. I just have Asterisk with config files |
| 23:55.06 | K0HAX | s/macro/goto |
| 23:57.09 | K0HAX | if you wanted the ability to add more, I'd recommend replacing the second dial line in mine with a goto to go to extension 1235 in that context, and basically duplicate extension 1234. That way you can add as many as you want. |
| 23:57.54 | K0HAX | (also, I would make the extension number something useful, like the trunk's phone number) |
| 23:59.01 | ruben23 | ok actually the scenario is there is 1 asterisk cloud peered to a main freePBX system...all the extension of cloud dials in then pass to freePBX and the freePBX dialsout for it |
| 23:59.23 | ruben23 | wanted to have a failover on it when the peers between this asterisk cloud and freepBX gets cut off |
| 23:59.39 | ruben23 | so the cloud asterisk can dialout by his own VoIP settign registered |
| 23:59.50 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |