| 00:00.10 | viasanctus | (to continue my statement of the 2nd vm) |
| 00:00.52 | viasanctus | we have a 100Mbs dedicated fiber connection with sip trunking |
| 00:00.55 | phix | cloud :/ I hate that term, IT IS JUST A VIRTUAL MACHINE! something that has existed since 1980 or something |
| 00:01.15 | viasanctus | phix, true, but a cloud offers more uptime |
| 00:01.19 | viasanctus | well..should |
| 00:01.26 | phix | cloud is just another buzz word |
| 00:01.36 | viasanctus | ah you mean that |
| 00:01.39 | viasanctus | yeah.. |
| 00:01.44 | phix | Developed by consumer research and marketing |
| 00:02.29 | viasanctus | i didn't want to go for an ha cluster |
| 00:02.54 | viasanctus | but something that doest reboots the vms on storage/computing failure of one of the nodes |
| 00:03.27 | viasanctus | in any case, I don't see why a 2nd vm in cluster with the first would scale out asterisk if its based on the same host |
| 00:09.54 | pabelanger | viasanctus, well, for me, the whole reason for using kvm is there no limit of resources I need to working about. We drop a proxy in front and load balance across them, if my instance is getting hammered and I need more, I simple spin up another. Once I don't need it any more, I destroy it and the resources are free to go back into the pool |
| 00:10.25 | pabelanger | however you do it, you should plan to make the instance dynamic / automated. So if one fails, you can quickly spin up a replacement |
| 00:10.58 | phix | pabelanger: what do you use to cluster kvm? a third party / commerical application? or just tedious scripting and configuration? |
| 00:11.08 | viasanctus | I think cloudstack has live failover mechanism for that |
| 00:11.21 | pabelanger | we are building upon OpenStack |
| 00:11.25 | viasanctus | ugh |
| 00:11.29 | viasanctus | why not cloudstack? |
| 00:11.44 | viasanctus | man I hate openstacks docs |
| 00:11.53 | pabelanger | No reason specific, we just use OpenStack |
| 00:12.02 | pabelanger | why do you use Asterisk and not Freeswitch |
| 00:12.06 | pabelanger | they both do the same thing |
| 00:12.08 | viasanctus | well, it's the same principle baiscally |
| 00:12.10 | viasanctus | indeed |
| 00:12.13 | pabelanger | just a matter of preference |
| 00:12.39 | viasanctus | openstack has some major companies backing it up..even the most beloved nsa I believe :) |
| 00:13.27 | pabelanger | yes, there are some smart people there too |
| 00:13.29 | phix | ah nsa |
| 00:13.47 | phix | and NASA |
| 00:13.52 | viasanctus | pabelanger, do you of use cases where the stack manages over 1000s of voip calls ? |
| 00:14.17 | pabelanger | viasanctus, ask me again in 3 months |
| 00:14.21 | viasanctus | lol |
| 00:14.31 | viasanctus | how so enthousiastic? :p |
| 00:14.46 | pabelanger | were are currently building out a new call centre, and starting the integration next week |
| 00:14.46 | phix | haha bluehost uses it, I just signed up with them :) |
| 00:14.52 | viasanctus | what hardware you running? |
| 00:15.09 | viasanctus | checks bluehost |
| 00:15.10 | pabelanger | Dell R515 are our compute nodes |
| 00:15.20 | pabelanger | cannot remember what the controller is ATM |
| 00:15.41 | viasanctus | object storage? |
| 00:15.55 | pabelanger | swift |
| 00:16.19 | viasanctus | local storage on the Dell R515 ? |
| 00:16.38 | pabelanger | Oh, no, we have some hardware coming in for that too |
| 00:16.46 | pabelanger | don't have the info in from of me |
| 00:17.03 | pabelanger | plan is to build out for live migration |
| 00:17.07 | pabelanger | but p1 will not support it |
| 00:17.29 | viasanctus | I often hear some professionals prefering local storage |
| 00:17.35 | viasanctus | and being able to scale out easily |
| 00:17.37 | pabelanger | but haven't tested asterisk with anything like that yet |
| 00:17.44 | viasanctus | your computing has to be linear with the storage though |
| 00:18.02 | viasanctus | if you're a storage provider, that'll probably be not the best way to go though |
| 00:18.04 | pabelanger | well, as long as you are find with your local node going down... hopefully the data is else ware in the cloud |
| 00:18.24 | viasanctus | it should be mirrorred around |
| 00:18.29 | viasanctus | one of the things I wonder about though |
| 00:18.50 | pabelanger | Also, right now we don't have heavy storage requirements right now. Just dealing call queues, recordings will be 6months out |
| 00:18.51 | viasanctus | it's a cloud setup in the end, but the mirrorring must consume a shitload of ethernet bandwidth |
| 00:19.00 | pabelanger | and, we might just use some external hardware for it |
| 00:19.22 | viasanctus | haven't heard of anyone using netapp with one of the stack sols |
| 00:19.31 | viasanctus | whilst it's very popular in vmware ha clusters |
| 00:19.46 | viasanctus | cephs is quite popular, but I don't think of going for objects |
| 00:20.35 | viasanctus | 16 opteron cores is smart thing to go for if you're using RHEL |
| 00:20.39 | viasanctus | are you? |
| 00:21.13 | pabelanger | are I what? |
| 00:21.18 | pabelanger | 16 cores or RHEL? |
| 00:21.25 | viasanctus | RHEL |
| 00:21.38 | pabelanger | centos 6.4 right now |
| 00:21.50 | pabelanger | but testing debian also |
| 00:21.55 | viasanctus | is it supported? |
| 00:22.01 | pabelanger | not sure which we'll use for baremetal |
| 00:22.07 | pabelanger | all our instances are ubuntu 12.04 |
| 00:22.10 | viasanctus | thought it was officially only supported by RHEL |
| 00:22.21 | pabelanger | every os is support |
| 00:22.36 | pabelanger | the amount of people using it for different OS's is different |
| 00:22.47 | pabelanger | not sure if RHEL offers commercial support yet or not |
| 00:22.55 | pabelanger | but, doubt we pay for it |
| 00:23.11 | viasanctus | RHEL is the only distro that does i believe |
| 00:23.36 | pabelanger | no |
| 00:23.53 | pabelanger | OpenStack will run on any Linux distro |
| 00:24.07 | viasanctus | I'm talking about kvm |
| 00:24.13 | viasanctus | commercial support |
| 00:24.21 | pabelanger | oh, have no idea |
| 00:24.36 | viasanctus | well doesn't matter, happy to hear another success story with centos |
| 00:24.47 | viasanctus | not willing to pay 2K$ for 2 sockets |
| 00:24.58 | pabelanger | well, we are only using centos to test out RDO |
| 00:25.02 | TriJetScud | CentOS might not be the choice for heavily threaded workloads because of a bug in libc/kernel |
| 00:25.07 | viasanctus | the whole reason I'm getting away from vmware |
| 00:25.11 | pabelanger | we have our own puppet modules, but wanted to try RDO |
| 00:25.13 | TriJetScud | FreeSWITCH has issues with CentOS 6.4 or below |
| 00:25.28 | TriJetScud | viasanctus: is vmware costing too much for you guys? |
| 00:25.38 | viasanctus | TriJetScud, yes |
| 00:25.53 | TriJetScud | heh, it's not the first time I've heard of that gripe |
| 00:25.57 | viasanctus | the HA + vcloud + vspp licenses are a killer |
| 00:26.08 | viasanctus | farting costs money per fart |
| 00:26.39 | viasanctus | you pay per connection per GB, per socket, per whatever is in the setup |
| 00:26.53 | viasanctus | you are obligated to buy yearly support |
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| 00:27.25 | viasanctus | they're pushing the whole market towards stacks |
| 00:27.39 | viasanctus | even xen has released their xenserver |
| 00:28.05 | phix | Centos :/ Debian and Ubuntu both exist! |
| 00:28.08 | viasanctus | whilst vsphere requires an essentials license to unlock hte agent if you wish to do anything more than hosting vms |
| 00:30.02 | viasanctus | TriJetScud, what distro would you recommend? |
| 00:36.04 | TriJetScud | viasanctus: debian based distros |
| 00:36.45 | TriJetScud | day to day for server use, I just go with debian, but for toying around, I just choose the latest ubuntu lts server edition |
| 00:37.27 | TriJetScud | but viasanctus, if you want to have the option of commercial support, you might want to look into microsoft's hyper-v |
| 00:38.04 | viasanctus | it'll most probably be kvm or xenserver |
| 00:38.13 | viasanctus | I'm done with properiatary |
| 00:38.18 | viasanctus | propriatary* |
| 00:39.16 | TriJetScud | viasanctus: well microsoft's hyper-v isn't all that bad, just think of Microsoft's own xen without the million of pages of documentation :P |
| 00:39.53 | viasanctus | not even sure cloudstack supports it |
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| 00:41.03 | viasanctus | thing with propriatary is that when you get into service providing and datacenter setups you always violate licenses in one or another way |
| 00:41.26 | viasanctus | can't image m$ will differ |
| 00:46.43 | TriJetScud | viasanctus: actually with openstack, microsoft's been supporting it to get their hyper-v platform to be a first class citizen |
| 00:47.08 | TriJetScud | and the free hyper-v server's EULA allows you to rent out VM's using their software |
| 00:47.34 | viasanctus | hmmm |
| 00:47.42 | viasanctus | i'll read upon it |
| 00:47.52 | viasanctus | although I'm not convinced of openstack |
| 00:48.07 | viasanctus | the central management gui of cloudstack is impressive |
| 00:48.17 | viasanctus | whilst dashboard of openstack frightened me |
| 00:48.46 | viasanctus | it feels less mature |
| 00:48.57 | viasanctus | and the whole stack does'nt seem as integrated |
| 00:49.16 | viasanctus | more a fragmented bunch of stuff that is agressively being developed |
| 00:49.23 | viasanctus | and bad documented |
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| 00:55.56 | pabelanger | viasanctus, we don't use the GUI, mostly use salt-cloud to provision our VMs |
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| 03:24.45 | hfp | Hi guys! I have a scenario where a user rings a DID number. This number doesn't pick up and calls the user back. The user is then provided with a dialtone to dial. The user then has the option of pressing *5 to record the call.This is my features.conf and relevant dialplan part: http://chopapp.com/#l2ze6eqo |
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| 03:25.58 | hfp | The location of the recording along with other arguments are passed to a bash script that transcodes the file to mp3 and emails it along with timestamps. Now, my problem is that I would like to play a "beep" or some kind of audio feedback to the caller that the recording has been activated. |
| 03:26.13 | hfp | But if I do this with Playback() in the macro, both parties hear it. |
| 03:26.47 | hfp | If I wanted only the user to hear the beep then I would have to change the features to be applied only to self and it won't record the other part of the call. |
| 03:27.25 | hfp | How could I record both parties but only play the beep to the user? |
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| 04:18.33 | ChannelZ | Why not just use a cough sound |
| 04:18.39 | ChannelZ | then the other party won't be the wiser |
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| 04:42.51 | hfp | True but I'd rather have a beep or something similar if at all possible, it's more professional |
| 04:44.27 | ChannelZ | BORING |
| 04:52.12 | jeev | hfp, ten horse sized ducks or one hundred duck sized horses? |
| 05:07.37 | lorsungcu | a horse sized duck screw penis would be terrifying. i will take the horses. |
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| 06:03.46 | phix | lorsungcu: heh |
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| 08:37.36 | giucio | I'd like to call ChanSpy through AMI on an existing channel |
| 08:38.02 | giucio | but I cannot find any reference to AMI commands implementing ChanSpy |
| 08:38.13 | giucio | does it exist? |
| 08:48.26 | ChannelZ | Hmm. I don't think so. |
| 08:51.19 | ChannelZ | though I'm not sure how that would even work if you could.. the spyer has to have a channel as well to listen, in what state would that person be in such that you could trigger a ChanSpy from AMI (or want to) |
| 08:52.40 | giucio | ChannelZ: OK then it might not be appropiate. What I'm trying to achieve, is to play a beep over an existing channel |
| 08:53.09 | giucio | I was under the impression that ChanSpy would do the trick |
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| 09:01.26 | ChannelZ | I don't know of a good way to do that, triggered externally |
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| 09:08.11 | eZz | hello, is there any issues with 'include' an additional conf files to musiconhold.conf ? I did an include but do not see any effect |
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| 09:13.59 | ChannelZ | Never tried with musiconhold specifically but I think it should work across 99% of the config files. |
| 09:14.23 | eZz | even I do not see Parsing... when moh reload |
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| 09:14.37 | eZz | and no new classes found after reload or a whole restart asterisk |
| 09:15.23 | ChannelZ | works for me. |
| 09:15.32 | ChannelZ | #include /etc/asterisk/moh2.conf |
| 09:16.36 | ChannelZ | you might have a syntax error elsewhere in your main musiconhold.conf or possibly even permissions |
| 09:17.13 | eZz | what version is your asterisk ? |
| 09:17.25 | eZz | I just tried at 1.8.23.1 and I got: [Sep 13 12:16:48] ERROR[17281]: config.c:1290 process_text_line: The file '=> /etc/asterisk/pdr.conf' was listed as a #include but it does not exist. |
| 09:18.33 | ChannelZ | 11.5 |
| 09:18.57 | eZz | so it can be just not implemented at 1.8 branch ? |
| 09:19.43 | ChannelZ | no it's obviously trying to read it, but you either type-o'd the name and it really doesn't exist, or maybe it's a permissions problem with the file (dunno if it generates a different error in that case or not) |
| 09:20.42 | eZz | nope, the file is exists, perms are ok |
| 09:21.39 | eZz | *CLI> !head -1 /etc/asterisk/pdr.conf |
| 09:21.39 | eZz | [1afa9f583624c510051e5f778c664eef] |
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| 09:27.30 | ChannelZ | Dunno, check you didn't mash an invisible control character or something invisible into the #include line I guess. |
| 09:28.13 | ChannelZ | I see an old bug about quoting the filename causing problems but don't think it applies here |
| 09:29.05 | eZz | I can't imagine how to add any invisible chars using vim :-] |
| 09:31.38 | eZz | I just copied this include to sip and I got: [Sep 13 12:30:51] ERROR[17239]: chan_sip.c:28589 reload_config: Contents of sip.conf are invalid and cannot be parsed |
| 09:31.46 | eZz | hence sip.conf's include was parsed |
| 09:32.11 | eZz | so there is definetely problem with parsing moh's includes I think |
| 09:36.25 | giucio | eZz: check with od -c the config file, just to rule out any weird chars possibility |
| 09:37.09 | ChannelZ | well the part that handles reading the config file should be the same function I think but not sure. Pastebin your configs.. my own test worked in 11, I've nothing running 1.8 to try there |
| 09:38.27 | eZz | giucio: http://pastebin.com/kUTRiynL |
| 09:39.56 | eZz | even it is doesn't works with http://pastebin.com/aJgPVsDD |
| 09:40.29 | eZz | I'm sure there is no invalid chars inside this file |
| 09:44.33 | ChannelZ | and your #include line in musiconhold.conf? |
| 09:44.38 | eZz | sure |
| 09:45.44 | eZz | ChannelZ: http://pastebin.com/wZMGaw9G |
| 09:46.16 | eZz | there is default musiconhold.conf with just one include at the bottom of file |
| 09:46.28 | ChannelZ | it's not #include => file |
| 09:46.33 | ChannelZ | remove the => |
| 09:46.41 | eZz | dang :> |
| 09:46.58 | eZz | <PROTECTED> |
| 09:47.12 | ChannelZ | If I was awake I'd have seen that in your initial question. Almost 4am, I gotta go to bed |
| 09:47.14 | eZz | yeah, need to sleep enough, guys ツ |
| 09:47.19 | eZz | thank you |
| 09:47.42 | ChannelZ | yup 'night |
| 09:47.42 | eZz | and with Programmer's Day everyone ! |
| 09:48.48 | eZz | 27 hours uptime makes a stupid mistakes |
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| 10:24.45 | gavimobile | I am having trouble diagnosing why one of my peers cannot connect. asterisk cli doesn't log any information about my remote peer. for testing purposes, I tried a softphone from my remote location and I was able to register. than I took this remote peer (not the softphone) and connected it to a different wan and this worked |
| 10:25.06 | gavimobile | so why is my peer unable to connect from the orignal remote location |
| 10:38.15 | gavimobile | all set.. apparently my peer was getting a dhcp but once I forced it a new dhcp in the router it started working |
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| 10:42.04 | BorjaGVO | Hello! |
| 10:42.51 | BorjaGVO | Anyone knows how can I make the following? |
| 10:43.03 | BorjaGVO | I'm extension 3333 (for example) |
| 10:44.16 | BorjaGVO | I want to call to someone and that inmediatelly after he picks up the phone, a recording is played to him. After it, return to me and be able to talk to him |
| 11:10.45 | Nickinator | Sure, |
| 11:10.53 | Nickinator | Let's call your extension you want to call 2222, |
| 11:10.59 | Nickinator | You'd set up your dialplan like this: |
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| 11:23.36 | eject_ck1 | Hi all |
| 11:23.58 | eject_ck1 | Why I see [Sep 13 14:22:28] WARNING[12174]: translate.c:162 framein: no samples for g729tolin in console + loosing 1-2 seconds at the very beginning of each call ? |
| 11:25.50 | jozza | whats the best way to pad a string to a certain length with zeros in a dialplan? |
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| 13:27.16 | sadtimes | hey |
| 13:27.30 | sadtimes | bit of a silly question; but I've been looking around and I can't find an answer |
| 13:27.35 | sadtimes | so perhaps someone can enlighten me |
| 13:27.51 | sadtimes | why if you set your rtp.conf to use a certain subset of ports |
| 13:28.13 | sadtimes | do you still see RTP going to a variety of ports outside of that range? |
| 13:28.36 | [TK]D-Fender | sadtimes: because rtp.conf are the ports YOUR system recieves on. |
| 13:28.51 | [TK]D-Fender | sadtimes: You send to where the OTHER side wants you to send to. |
| 13:28.52 | aelliott22 | depends on the other end |
| 13:29.04 | igcewieling | sadtimes: prove it. I've never seen the Asterisk side using ports outside that range. Of course Asteirsk can't control what remote devices choose for thei source ports. |
| 13:30.32 | igcewieling | People doing VoIP and not understanding basic TCP/IP concepts is going to be the death of VoIP |
| 13:30.52 | sadtimes | igcewielding: okay? |
| 13:31.10 | sadtimes | pretty sure you have 0 knowledge of what my understanding of TCP/IP is |
| 13:31.22 | igcewieling | sadtimes: I've seen a couple of similar questions over the past few weeks. The poor sods wasted days of their time because of this. |
| 13:31.23 | [TK]D-Fender | igcewieling: Down boy! |
| 13:31.30 | sadtimes | and being a complete ass to people who join channels is not the greatest way to build communities |
| 13:31.54 | igcewieling | sadtimes: I'm sorry if I came across as mean. |
| 13:32.30 | sadtimes | obviously the src and dst are going to be different |
| 13:33.06 | Greenlight | The point he's making is that you only have control over the port at YOUR side, not the REMOTE side |
| 13:33.16 | sadtimes | yeah know I know that |
| 13:33.20 | sadtimes | balls |
| 13:33.27 | sadtimes | meant to say yeah no I know that |
| 13:35.28 | Katty | GOOD MORNING CUPCAKES |
| 13:35.30 | Katty | IT"S FRIDAY! |
| 13:37.03 | _Corey_ | checks calendar carefully. |
| 13:37.51 | Greenlight | We're only open Thursdays... |
| 13:39.00 | Katty | no sure if George Carlin fan, or just has seen me yell random things into the channel... |
| 13:39.13 | Greenlight | ^^ |
| 13:39.50 | Greenlight | I only know of George Carlin from your random shouts :) |
| 13:39.58 | Katty | nods |
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| 13:43.07 | ACiDV | Hi, I want to write an extern with pattern matching like : exten => _[dD][nN][dD][-_].,NoOp(${EXTEN:4}) .... so I can trap dnd-1234, DND_1234, Dnd-1234 etc... it trap correctly the underscore but not when I use key with a dash -, does [-_] is a valid syntax ? |
| 13:43.34 | sadtimes | gah sigh I think I was just being retarded |
| 13:43.35 | sadtimes | so tired :/ |
| 13:44.37 | bacobart | ehm im not sure if this is true in the dialplan but you might need to escape the dash |
| 13:45.12 | ACiDV | bacobart: yes I tried to use [\-_] but failed, continue to check |
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| 13:47.59 | ACiDV | In this case I presume I will have to have 2 rules, one for - and the other for _ |
| 13:48.00 | Katty | sadtimes: naptime. |
| 13:48.21 | [TK]D-Fender | ACiDV: Do they work independently? |
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| 13:48.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:48.31 | Katty | waves to sruffell |
| 13:48.50 | ACiDV | [TK]D-Fender: if I set a rule for _[dD][nN][dD]- and another for _[dD][nN][dD]_ it work |
| 13:48.57 | Qwell | hrm |
| 13:49.01 | [TK]D-Fender | ACiDV: also I know that "-" is used in [] for ranges, so maybe swapping the order might help |
| 13:49.01 | Qwell | aren't dashes ignored? |
| 13:49.09 | [TK]D-Fender | Qwell: Possibly... |
| 13:49.46 | ACiDV | If I set [_-], asterisk cli complain about a missing ] |
| 13:49.47 | kchehab | i am using a realtime sip and iax users, my asterisk 11.5 was working fine and suddenly i get this error in debug file WARNING[2666] config.c: Realtime mapping for 'iaxpeers' found to |
| 13:49.47 | kchehab | <PROTECTED> |
| 13:49.56 | kchehab | how can i fix odbc |
| 13:50.09 | Qwell | [-_] is fine, but, I don't think it comes into the dialplan containing the -. |
| 13:50.34 | Katty | hugs Qwell |
| 13:50.56 | Qwell | is attacked |
| 13:51.39 | ACiDV | Qwell: [-_] doesn't generate error but - isn't matched, probably that it a "limitation" of pattern matching on * :) |
| 13:51.51 | Qwell | It's not the fault of the pattern. |
| 13:52.02 | Qwell | I'm saying that the item you're matching does not actually contain a -. |
| 13:53.08 | *** join/#asterisk P424D0X (~kvirc@dslb-188-102-020-108.pools.arcor-ip.net) |
| 13:54.14 | Qwell | exten => _[dD][nN][dD]_.,1,Goto(DND${EXTEN:4}) |
| 13:54.17 | ACiDV | Qwell: for rule : exten => _[dD][nN][dD][-_].,NoOp(...) .... If I dial " dnd-1234 " and fail, but if I dial " dnd_1234 " so it contain a - ? |
| 13:54.21 | Qwell | exten => _[dD][nN][dD].,1,Magic(!) |
| 13:55.30 | P424D0X | Who can help me out?! My asterisk can't register to 1&1 and sipgate.. Username and passwords are correct.. Firewall will forward SIP traffic.. |
| 13:55.47 | ACiDV | Qwell: ok thanks will check, so cannot use a single rule to match all case |
| 13:55.57 | kchehab | Qwell ho :) |
| 13:56.30 | kchehab | hi |
| 13:57.04 | [TK]D-Fender | P424D0X: Show us the SIP debug from * CLI of your attempts. |
| 13:57.09 | [TK]D-Fender | ~pb |
| 13:57.10 | 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. |
| 13:57.11 | [TK]D-Fender | ^^^ |
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| 13:57.31 | *** join/#asterisk paule32 (~paul@dslb-188-106-252-082.pools.arcor-ip.net) |
| 13:57.36 | paule32 | hello |
| 13:57.54 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 13:58.25 | paule32 | how can i set the port in sip.conf? |
| 13:58.45 | [TK]D-Fender | paule32: bindport=THEPORT |
| 13:58.51 | [TK]D-Fender | paule32: under [general] |
| 13:59.04 | [TK]D-Fender | paule32: do read the sip.conf sample config.. it's all in there |
| 13:59.14 | paule32 | because, i use a softphone (zoiper) and can't register |
| 13:59.24 | paule32 | how to login? |
| 13:59.35 | paule32 | 1001@localhost ? |
| 14:00.04 | paule32 | [TK]D-Fender, thx |
| 14:00.12 | [TK]D-Fender | paule32: user & pass are separate fields there last I checked... |
| 14:00.56 | [TK]D-Fender | [09:59]paule32because, i use a softphone (zoiper) and can't register <- you should probably be showingf us your failed attempts |
| 14:01.12 | paule32 | or in other words, how is the domain to login in intranet? |
| 14:01.17 | paule32 | mom |
| 14:01.44 | [TK]D-Fender | paule32: ...? |
| 14:03.22 | P424D0X | [TK]D-Fender: Here is the output of debug: http://pastebin.com/1TWnEZuh |
| 14:04.53 | [TK]D-Fender | P424D0X: there is no registration attempt in there at all |
| 14:05.20 | kchehab | is it an ODBC problem ? |
| 14:05.26 | P424D0X | [TK]D-Fender: Hmm.. Why?! :-( |
| 14:05.50 | P424D0X | [TK]D-Fender: Do you have an advice?! |
| 14:06.55 | [TK]D-Fender | P424D0X: I don't know why. maybe you didn't wait long enough. You didn't issue a "sip reload" to try to force it to re-register. Maybe you didn't configure sip.conf properly to register at all... |
| 14:07.10 | [TK]D-Fender | P424D0X: Could be any of several different reasons. |
| 14:07.55 | [TK]D-Fender | kchehab: That warning seemed pretty clear to me... |
| 14:08.27 | sruffell | morning Katty |
| 14:08.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:08.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.07 | P424D0X | [TK]D-Fender: But that hasn't to do with dial plan?! |
| 14:09.19 | [TK]D-Fender | P424D0X: No. |
| 14:13.02 | paule32 | [TK]D-Fender, http://imageshack.us/f/844/cunw.png/ |
| 14:13.37 | P424D0X | [TK]D-Fender: In file sip.conf I have a entry "#include sip_registrations.conf" and in file sip_registrations.conf I have two lines like "register => AAA:BBB@sip.1und1.de/AAA".. |
| 14:14.21 | bacobart | paule32: what softphone is that? |
| 14:14.37 | paule32 | bacobart, zoiper |
| 14:14.41 | bacobart | thanks |
| 14:15.22 | kchehab | [TK]D-Fender is this error: WARNING[2666] config.c: Realtime mapping for 'iaxpeers' found to |
| 14:15.22 | kchehab | <PROTECTED> |
| 14:15.36 | kchehab | is an odbc error |
| 14:16.00 | Qwell | P424D0X: Where is the #include? |
| 14:16.28 | paule32 | bacobart, have you a tip how to connect sip phones with local networks? |
| 14:16.51 | P424D0X | Qwell: It's in sip.conf.. general part |
| 14:16.59 | Qwell | show us |
| 14:17.35 | [TK]D-Fender | P424D0X: I also told you 1 thing you could do to trigger a re--registration.... |
| 14:18.47 | [TK]D-Fender | paule32: what is the IP of your windows PC? |
| 14:19.09 | bacobart | paule32: im not sure, but seems to me your pc can't reach the sip server (timeout) |
| 14:19.13 | bacobart | as to why i have no idea |
| 14:19.17 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 14:24.30 | paule32 | bacobart, win 192.168.178.79, linux vm 192.168.178.200 |
| 14:24.53 | paule32 | but maybe the config is not fine for vm box |
| 14:25.06 | paule32 | because i use a bridged network |
| 14:25.24 | [TK]D-Fender | paule32: verify that you are in bridged LAN mode for your VM, and check your firewall on your *NIX side |
| 14:25.25 | paule32 | that is correspond to lan nic realtek |
| 14:25.39 | [TK]D-Fender | paule32: No, it needs to be bridged, not NAT'd |
| 14:26.00 | paule32 | yes bridged is ok |
| 14:26.00 | [TK]D-Fender | paule32: Go check your firewall.. then go see if * is even listening for SIP |
| 14:26.19 | [TK]D-Fender | paule32: netstat -an|grep 5060 |
| 14:28.10 | paule32 | root@debian /etc/asterisk > netstat -an | grep 5060 |
| 14:28.11 | paule32 | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 14:28.38 | P424D0X | [TK]D-Fender: Qwell: Okay.. Here is my complete sip configuration: http://pastebin.com/XLHn5x9u |
| 14:28.54 | [TK]D-Fender | paule32: check your firewall |
| 14:29.16 | paule32 | on linux side? |
| 14:29.20 | [TK]D-Fender | paule32: BOTH |
| 14:29.37 | [TK]D-Fender | paule32: Windows could block Zoiper direct, and Linux as well |
| 14:29.54 | [TK]D-Fender | P424D0X: "sip reload" <--- show an actual registration. |
| 14:30.15 | [TK]D-Fender | P424D0X: and "sip show registry" while you're waiting... |
| 14:31.05 | hjf | hehe too bad igcewieling isn't here |
| 14:31.30 | P424D0X | Host dnsmgr Username Refresh State Reg.Time |
| 14:31.30 | P424D0X | sipgate.de:5060 N 2473867 300 Unregistered |
| 14:31.30 | P424D0X | sip.1und1.de:5060 N 490384173846 300 Unregistered |
| 14:32.59 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 14:34.07 | P424D0X | When I enter "sip reload" and "sip show registry", then it will output nothing.. Only when I logout from CLI and login to CLI of asterisk.. |
| 14:34.28 | kchehab | how can i fix asterisk ODBC ? |
| 14:34.51 | [TK]D-Fender | kchehab: I see nothing to indicate that * is the problem. |
| 14:36.15 | kchehab | [TK]D-Fender i cant find odbc in asterisk -r |
| 14:36.42 | kchehab | [TK]D-Fender this erros means that the problem is in odbc ? |
| 14:37.48 | [TK]D-Fender | kchehab: I have no idea what you actually did or mean by "i cant find odbc in asterisk -r". You haven't proven that ODBC is functional outside of *, we haven't seen configs, nor attempts to load/reload the module from CLI. You haven't shown us anything for us to help you with. |
| 14:38.49 | kchehab | i can just find this error in asterisk/full WARNING[2666] config.c: Realtime mapping for 'iaxpeers' found to engine 'odbc', but the engine is not available |
| 14:40.41 | *** join/#asterisk monsterco (63f3a3f2@gateway/web/freenode/ip.99.243.163.242) |
| 14:41.45 | monsterco | Hi everyone - I was here asking for advice on Polycom phones yesterday - I have two sets that might be provisioned but are not locked - Can I proceed with a File System Format and expect it to Factory default *ALL* the settings? |
| 14:42.16 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 14:42.23 | monsterco | *Format File System* |
| 14:42.41 | [TK]D-Fender | monsterco: Go into the bootrom, kill the provisioning server settings THEn do a factory reset |
| 14:44.26 | wdoekes | kchehab: *CLI> module reload res_odbc.so |
| 14:44.45 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:47.09 | kchehab | wdoekes No such module 'res_odbc.so' |
| 14:47.39 | kchehab | wdoekes even i cant do module load res_odbc.so |
| 14:47.49 | kchehab | but i can find it in module directory |
| 14:48.27 | wdoekes | if you remove it and re-do make install, does it reappear? or are you looking at an old module? |
| 14:49.22 | kchehab | wdoekes i will check it |
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| 15:01.41 | *** join/#asterisk monsterco (63f3a3f2@gateway/web/freenode/ip.99.243.163.242) |
| 15:02.32 | monsterco | [TK]D-Fender - Ok, so I am with the phone now. By disabling Provisioning server you mean reboot the phone, go into setup and then REMOVE the SERVER ADDRESS right? |
| 15:02.38 | monsterco | or is there anything else I have to do? |
| 15:02.46 | [TK]D-Fender | no. |
| 15:03.14 | monsterco | I have a lot more patience today and am on site so let's start step by step :) |
| 15:04.01 | monsterco | TK]D-Fender - so how do I disable provisioning? |
| 15:04.26 | [TK]D-Fender | That *IS* how. |
| 15:04.26 | WIMPy | The two day battle with one phone. |
| 15:04.53 | monsterco | lol |
| 15:05.25 | monsterco | [TK]D-Fender- well, i will waste 2 hours here and I have Aastra phones to replace if the garbage thing doesn't work |
| 15:06.06 | [TK]D-Fender | It works |
| 15:06.15 | monsterco | so what is my next step here? |
| 15:06.33 | [TK]D-Fender | kill the provisioning server. Factory Default. Reconfigure |
| 15:07.15 | monsterco | how do I kill the provisioning server? remove ethernet cable? |
| 15:07.26 | [TK]D-Fender | ... |
| 15:07.34 | [TK]D-Fender | ERASE that entry. The same thing I've told you a dozen times |
| 15:07.43 | [TK]D-Fender | Enter bootrom. REMOVE the server address |
| 15:07.45 | [TK]D-Fender | THE END |
| 15:09.40 | *** join/#asterisk paule32 (~paul@dslb-188-106-252-082.pools.arcor-ip.net) |
| 15:09.54 | paule32 | oops sorry, connection reset |
| 15:09.59 | paule32 | is this zoiper domain correct: "sip:1001:1234@192.168.178.200" ? |
| 15:10.35 | [TK]D-Fender | paule32: lthat is not a "domain", and we have no idea where you are trying to put that "value" |
| 15:10.55 | [TK]D-Fender | paule32: You were supposed to verify your firewalls and come back to us |
| 15:11.32 | WIMPy | wonders what a zoiper domain might be. |
| 15:16.09 | paule32 | ok, firewalls ok, no blocking |
| 15:16.25 | *** join/#asterisk monsterco (~monsterco@CPE000db91f04bd-CM001bd7092ef6.cpe.net.cable.rogers.com) |
| 15:16.35 | paule32 | but telnet localhost 5060 refused |
| 15:16.48 | monsterco | [TK]D-Fender - sorry network issues here |
| 15:16.53 | paule32 | also no listener on port 5060 |
| 15:17.00 | monsterco | I removed the Server Address from BOOTROOM |
| 15:17.04 | monsterco | what is my next move now? |
| 15:17.59 | [TK]D-Fender | [11:06]monstercoso what is my next step here? [11:06][TK]D-Fenderkill the provisioning server. Factory Default. Reconfigure |
| 15:18.42 | [TK]D-Fender | [11:16]paule32but telnet localhost 5060 refused <- SIP is ***UDP*** by default. Telnet = TCP and is not a valid test |
| 15:19.04 | paule32 | ok |
| 15:20.58 | monsterco | how do I Factory Default? |
| 15:21.13 | paule32 | in sip.conf [general] -> bindport=5060 tcpbindaddr=:: topenable=no ? |
| 15:21.45 | [TK]D-Fender | monsterco: I've also repeated this many times through yesterday that you came in HAVING that answer and telling us you've done it already. You KNOW the key comb ination already. |
| 15:22.04 | monsterco | this is a new set |
| 15:22.14 | monsterco | i am just making sure it's not a set issue |
| 15:23.06 | [TK]D-Fender | monsterco: The instructions are the same for the model... |
| 15:23.13 | monsterco | I am not sure what you mean by Factory Default - I did remove Server Address for Ring Central in BOOTROOM |
| 15:23.30 | monsterco | the phone is now restart - Updating initial Configuration it says |
| 15:25.52 | monsterco | Rev: 3.2.1.0078 now |
| 15:28.38 | monsterco | ok the phone is now booted |
| 15:28.43 | monsterco | what should I do now? |
| 15:29.11 | [TK]D-Fender | [11:17][TK]D-Fender[11:06]monstercoso what is my next step here? [11:06][TK]D-Fenderkill the provisioning server. Factory Default. Reconfigure |
| 15:30.00 | monsterco | What do you mean by "Factory Default" |
| 15:30.33 | [TK]D-Fender | monsterco: Reset to factory default |
| 15:30.48 | [TK]D-Fender | monsterco: with the key combination you already came in here showing us you knew |
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| 15:33.52 | monsterco | ok, so I will press 1,3,5,7 at countdown |
| 15:34.12 | monsterco | but I think that put the phone back into RingCentral provisioning again last I did - let me check now |
| 15:34.18 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6ccc:cafe:dead:0:beef) |
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| 15:34.55 | gmalsack | hey all ~ trying to minimize extensions.conf to only parts that I don't expect to change, and move stuff that will change often to sql. |
| 15:35.42 | *** join/#asterisk adeeln (~adeel@199.48.103.26) |
| 15:36.08 | gmalsack | I use variables to set the outgoing caller id of certain users. I want to get those global variables out of extensions.conf but not sure what the best choice is. astdb or sql? |
| 15:36.49 | monsterco | [TK]D-Fender> - so I held 1,3,5,7 at countdown and it didn't ask me for a password but it did say resetting configurations |
| 15:36.58 | monsterco | and it is now rebooted |
| 15:37.23 | monsterco | once rebooted I will reboot again to see if the provisioning server settings are back or not to make sure |
| 15:37.26 | paule32 | why is port 0 ?? http://codepad.org/ZSsvyI0j |
| 15:37.55 | [TK]D-Fender | paule32: Because the device has not registered and * has no IP to contact at all, let alone a port |
| 15:38.07 | [TK]D-Fender | paule32: So the "0" doesn't matter". * has nowhere to call. |
| 15:39.20 | paule32 | [TK]D-Fender, config wrong? |
| 15:39.55 | [TK]D-Fender | paule32: Not that I can see. So far I have no proof that an attempt ever ARRIVED at your server |
| 15:40.57 | paule32 | [TK]D-Fender, do you can tell me, how softphones are registered at asterisk? |
| 15:41.19 | [TK]D-Fender | paule32: I cannot understand your wording. Please rephrase |
| 15:42.12 | paule32 | how to register a softphone (sip) at the asterisk server |
| 15:43.00 | [TK]D-Fender | paule32: configure the softphoe to register. Have a peer setup on * that it can register. Make sure your networking is not interfering. |
| 15:44.04 | paule32 | is it enough to write: sip:user@ip ? |
| 15:44.58 | gmalsack | anyone have any input on storing global variables in astdb vs sql? |
| 15:45.59 | monsterco | this phone takes for ever to restart |
| 15:46.16 | gmalsack | monsterco: polycom? |
| 15:46.19 | WIMPy | Global variables are strored in RAM and neither in AstDB nor in any SQL. |
| 15:46.20 | monsterco | yeah |
| 15:46.57 | Katty | hello my phone is not working at all it does not ring how to fix plz? is urgent thx. |
| 15:47.19 | monsterco | [TK]D-Fender> - Now the phone is rebooted after I did 1,3,5,7. I am just on the WEB GUI and I see that all the line settings I put there are still there. Does that mean that the phone did NOT factory default? |
| 15:47.21 | Weezey | I have a user whose asterisk box is DMZ'd and when it registers it registers to my switch as the private IP instead of the public IP. |
| 15:48.01 | [TK]D-Fender | Weezey: So go fix the NAT settings in their sip.conf just like we've always had to do |
| 15:48.09 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 15:48.21 | Katty | i see fender is in a lovely mood this morning |
| 15:48.23 | monsterco | Weezey - localnet and externalip is what you are looking for |
| 15:48.28 | Weezey | [TK]D-Fender: cool, thanks |
| 15:48.31 | [TK]D-Fender | Weezey: DMZ is also a horrible idea. Forwardonly the required ports or you open yourself up to many more attack vectors |
| 15:48.42 | Katty | [TK]D-Fender: hows the healing going? |
| 15:48.51 | Weezey | [TK]D-Fender: temporary cellular setup, static IP coming next week |
| 15:49.13 | Katty | i never figured that out. it doesn't actually take a week to assign an IP |
| 15:49.36 | monsterco | [TK]D-Fender> - I don't think this phone has factory defaulted - it still shows the line settings I put in there - I just restart and went into BOOTROOM again and I see that RingCentral Server Address came back |
| 15:50.17 | [TK]D-Fender | monsterco: perhaps there is a DHCP provisioning option being pushed onto the phone. |
| 15:50.20 | monsterco | this happens when I do the 1,3,5,7 process at count down. so, it seems these phones are putting all the provisioning settings back in the phone when I do 1,3,5,7 process - or maybe I am doing that wrong |
| 15:50.26 | monsterco | option 66? |
| 15:50.51 | gmalsack | wimpy: i realize that.... my question is whether I'm better off moving the global variables to astdb or sql in regards to performance and ast payload... |
| 15:51.52 | WIMPy | gmalsack: No DB will outperform global variables. |
| 15:52.13 | [TK]D-Fender | monsterco: 66/150 |
| 15:52.46 | gmalsack | wimpy: these variables will change often. I don't want to reload my dialplan. so I'm trying to remove everything that will change from extensions.conf |
| 15:53.22 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 15:53.33 | WIMPy | gmalsack: So they are not variables, but constants? |
| 15:53.48 | *** join/#asterisk hdiogenes (~hdiogenes@187.60.66.11) |
| 15:53.57 | monsterco | [TK]D-Fender> - so I will disable DHCP and set static ip |
| 15:53.58 | *** part/#asterisk hdiogenes (~hdiogenes@187.60.66.11) |
| 15:54.19 | [TK]D-Fender | monsterco: go into the bootrom and make sure to kill off those DHCP features |
| 15:54.21 | WIMPy | And you don't have to reload dialplan to set global variables anyway. |
| 15:54.22 | monsterco | but there is no SIP server on local net so I doubt option 66 or 150 would lock the phone again |
| 15:54.22 | gmalsack | I was taught they are global variables.... |
| 15:54.26 | [TK]D-Fender | monsterco: those are ALSO in there |
| 15:54.43 | gmalsack | monster: here's a pastebin from my dhcp server for my polycom phones: http://pastebin.com/btjNqcgy |
| 15:54.46 | adeeln | anyone have any familiarity with voicexml & ivrs in asterisk? |
| 15:54.52 | monsterco | [TK]D-Fender> - yes it is but aren't those for local network? |
| 15:55.31 | WIMPy | gmalsack: Well, then you better tell us what exactely you're looking for instead of us guessing. |
| 15:55.34 | [TK]D-Fender | monsterco: normally yes. just kill all of those off, and make sure all of the other bootrom settings are flushed. |
| 15:57.07 | gmalsack | wimpy: here's a pastebin of what im talking about: http://pastebin.com/btjNqcgy |
| 15:57.46 | gmalsack | I want to move the CID_XXXX out of extensions.conf to either astdb or sql. just not sure which would be more efficient. |
| 15:57.52 | WIMPy | gmalsack: Your "global variables". |
| 15:58.08 | gmalsack | just the CID_XXXX ones |
| 15:58.20 | gmalsack | which was I correct? they are global variables? |
| 15:58.28 | WIMPy | gmalsack: Thqt doesn't mean anything to anyone other than yourself. |
| 15:58.45 | gmalsack | wimpy: how do you figure? |
| 15:58.50 | WIMPy | gmalsack: Are you talking about sip users? Then set the variables inthe peers. |
| 15:59.41 | gmalsack | wimpy: oh right.... I could set that on the peer configurations.... I forgot about that! THANK YOU VERY MUCH!!! |
| 15:59.43 | WIMPy | It's not as if "CID_XXXX" is anything standard. |
| 16:00.27 | gmalsack | really???? duh I know that. |
| 16:00.37 | gmalsack | i.e. called a variable.... |
| 16:01.00 | gmalsack | that's why I kept asking about moving the variable. |
| 16:01.14 | monsterco | [TK]D-Fender> - boot server i have Option 66, Custom+opt.66 and Static |
| 16:01.15 | gmalsack | but for some odd reason you couldn't wrap your head around that... |
| 16:01.26 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:01.31 | monsterco | when I disable DHCP and set static IPs ofcourse those don't matter - but that again doesn't work |
| 16:01.48 | monsterco | if I put Boot Server option to STATIC and do 1,3,5,7 that also put the ringcentral back in place |
| 16:01.53 | WIMPy | gmalsack: You original question was impossible to answer without knowing where and when the variable will be read or written. |
| 16:02.09 | monsterco | there is clearly something else that is being missed. Is it possible that this phone is still locked to RingCentral? |
| 16:02.29 | gmalsack | not really.. |
| 16:03.00 | *** join/#asterisk jhlavacek (~jirka@212.234.54.86) |
| 16:03.07 | monsterco | [TK]D-Fender> - what if I disconnect the ethernet cable - remove Server Address, and then do Factory default? |
| 16:03.08 | gmalsack | but thank you for helping. you did answer my question and get me where I wanted to go... |
| 16:03.20 | [TK]D-Fender | monsterco: removing the ethernet cable does nothing. |
| 16:03.42 | monsterco | p.s. and just to make sure I am in right place - by BOOTROOM you mean when the phone restarts and it does countdown and gives me SETUP option right? |
| 16:03.51 | [TK]D-Fender | monsterco: just kill the boot server and your manual settings should stick |
| 16:05.10 | *** join/#asterisk navaismo (~navaismo@189.241.77.253) |
| 16:05.57 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 16:09.25 | monsterco | [TK]D-Fender - it works for one reboot and then it reboots again and puts the RingCentral Server Address in there again |
| 16:09.42 | monsterco | there is something else that is being missed - or maybe the 1,3,5,7 option doesn't really work |
| 16:10.45 | sawgood | monsterco: by chance, are you working on a possible provider locked Polycom phone? |
| 16:12.30 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 16:12.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:12.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:16.25 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 16:21.36 | monsterco | sawgood - it's possible that the phone is locked to RingCentral but they told me it's not locked any longer |
| 16:21.41 | monsterco | how do I know if it's locked or not? |
| 16:21.48 | monsterco | has a logo of RingCentral on it too |
| 16:23.19 | [TK]D-Fender | monsterco: Tactical nuke option -> extract a stock firmware into a FTP folder and reprovisioning it locally including the bootrom and that will annihilate everything on the phone |
| 16:31.11 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 16:32.26 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:36.08 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 16:41.00 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 16:46.04 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 16:46.14 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 16:47.28 | Weezey | after enabling externhost the other peers could see me but I can't register anymore. Getting registration for x@host timed out, trying again |
| 16:47.35 | Weezey | dns lookup for host works fine |
| 16:49.03 | Weezey | maybe a co-incidence? but how can i debug it |
| 16:53.06 | [TK]D-Fender | Weezey: "sip set debug on" <- |
| 16:53.13 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.216) |
| 16:55.03 | Weezey | nothing glaring at me. maybe I need a port number in the router? |
| 16:55.49 | [TK]D-Fender | Weezey: I told you that DMZ= BAd and you should just forward the appropriate ports |
| 16:55.58 | Weezey | you did. |
| 16:56.29 | Weezey | but I'm not on site so it makes shit sticky, |
| 16:56.47 | Weezey | stupid router reboots after every change. |
| 16:57.40 | Weezey | it also gets a new IP after every reboot, for good measure. |
| 16:57.55 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 16:58.20 | Weezey | VoIP over LTE is a temporary solution to a slow truck roll, but nothing more. |
| 16:58.34 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 16:59.02 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 16:59.24 | paule32 | ok, sip phone is registered at server |
| 16:59.32 | paule32 | express talk |
| 16:59.37 | paule32 | nch |
| 17:00.31 | paule32 | call 1001 brings me an endless ring tone |
| 17:00.45 | paule32 | instead "say hello world" |
| 17:01.31 | navaismo | show us your dialplan |
| 17:02.22 | [TK]D-Fender | paule32: same => 0,Playback(hello-world) <- bad PRIORITY number |
| 17:02.28 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 17:02.43 | [TK]D-Fender | paule32: You answer.. and then do nothing because you run out of valid steps |
| 17:03.59 | paule32 | ah ok |
| 17:06.59 | paule32 | it works fine |
| 17:07.17 | paule32 | it is funny, that io can recall me |
| 17:07.50 | paule32 | but when i try to change the line the call is droped |
| 17:14.54 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ed0a:91a:d89a:d657) |
| 17:15.19 | Penguin | What does "change the line" mean? |
| 17:16.32 | paule32 | Penguin, phone channel |
| 17:17.05 | paule32 | express talk nch has 6 channels |
| 17:17.52 | paule32 | when i dial 1002, a friendly women voice will say hello world |
| 17:18.01 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 17:20.06 | karl-s | have any of you ever done a type of loopstart signalling over T1? Does Asterisk/DAHDI support hookflash features over that channel? |
| 17:21.04 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 17:22.11 | ChannelZ-Wk | (ot) does running MacOS on VirtualBox really work? |
| 17:22.29 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 17:23.41 | paule32 | when i call 1001 on a soft phone, hello world will speaking |
| 17:23.51 | jmetro | ChannelZ-Wk: probably, i have an image of macosx that worked on VMware |
| 17:23.53 | paule32 | but in the same time, asterisk console: |
| 17:23.55 | paule32 | [Sep 13 19:22:23] WARNING[16823]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 17:24.01 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 17:24.19 | jmetro | paule32: post your dialplan on a pastebin , you have something wrong that will crash the internet. |
| 17:24.59 | [TK]D-Fender | paule32: that is the typical error when you are dialing a peer that has not registered |
| 17:27.18 | paule32 | jmetro, http://codepad.org/bYtETU73 |
| 17:28.39 | [TK]D-Fender | paule32: exten => 1002,1,Dial(SIP/1002) exten => 1002,1,Answer() |
| 17:28.47 | [TK]D-Fender | paule32: that is the same extension & priority\ |
| 17:28.54 | [TK]D-Fender | paule32: You are overwriting your dialplan there |
| 17:29.49 | jmetro | has anyone realized that with Apple building a database of your fingerprints, youre never going to have to get fingerprinted by the government again since they'll already have your records? |
| 17:30.10 | ChannelZ-Wk | jmetro: huh. I knew they were always battling the Hackintosh crowd, didn't figure they'd let a VM work. |
| 17:31.48 | Weezey | aha! The router had SIP ALG on by default. |
| 17:31.53 | jmetro | ChannelZ-Wk: Well think of it like this, the OS is just a program that wants parameters, and you just feed it what its expecting |
| 17:32.02 | jmetro | Weezey: wah wah wah wahhhhh |
| 17:33.01 | Weezey | jmetro: I don't like it when things try to be smart. |
| 17:34.29 | jmetro | Weezey: i dont think i've ever seen sip ALG work. |
| 17:35.09 | ChannelZ-Wk | Right I just didn't know they were actively tricking it. I thought Apple liked to sue people |
| 17:36.38 | jmetro | dont know why anyone would bother running apple on anything that isnt a 3000$ facebook machine though |
| 17:37.41 | ChannelZ-Wk | although now since the OS is a download app.. does it leave behind a disc image or anything you could burn to boot from even on a real mac should you wipe out your old system? |
| 17:38.11 | jmetro | the best method is probably blank install -> acronis |
| 17:38.26 | jmetro | or .. whatever 100$ mac backup software you have to buy |
| 17:38.30 | ChannelZ-Wk | I just need it for the occasional client drive that someone brings in some f*ed up files on, our Mac spends most of its time running Windows these days |
| 17:38.54 | jmetro | oh you mean like a dead hard drive? plug a live-CD into that sucker |
| 17:40.32 | paule32 | ok sip phones can communicate |
| 17:40.56 | paule32 | what have i to do, to call outside the localnet |
| 17:42.10 | Penguin | Set up a peer that isn't on the LAN. |
| 17:42.16 | paule32 | something with "registrar"? |
| 17:42.20 | Penguin | No. |
| 17:42.29 | karl-s | jmetro, sometimes you have to |
| 17:42.46 | karl-s | regarding sip ALG i mean |
| 17:43.05 | karl-s | like when you have two ISP's -- Asterisk cant currently handle 2 public IP's... |
| 17:44.55 | jmetro | hm, you cant do a redirect? |
| 17:45.21 | file | chan_sip can't handle it |
| 17:45.25 | file | #pedantic |
| 17:48.22 | *** join/#asterisk blizzow1 (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 17:49.18 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 17:50.09 | *** join/#asterisk cwilson7938 (~benderUSP@nat/digium/x-adcmbaunopqgamtc) |
| 17:54.01 | Qwell | It's...a cwilson7938... |
| 17:54.15 | file | a wild one |
| 17:54.39 | Qwell | file: look what you went and did |
| 17:54.55 | file | yes, all my fault |
| 17:58.30 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.216) |
| 18:07.23 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:09.05 | jmetro | is operator=yes in voicemail.conf still the way to get an escape from the vm? |
| 18:09.11 | jmetro | doesnt seem to work for me. |
| 18:09.27 | [TK]D-Fender | jmetro: Show us what you configured |
| 18:09.57 | jmetro | also - does it work if there are no 0 or *'s defined. |
| 18:11.01 | *** join/#asterisk Alex36 (uid12585@gateway/web/irccloud.com/x-hfvdzhzjlilthbok) |
| 18:13.01 | jmetro | voicemail show usaers for Context gives me this.. http://pastebin.com/VQQqYp8a |
| 18:13.10 | Penguin | No, it requires extension o. |
| 18:14.08 | j4jackj | <PROTECTED> |
| 18:14.16 | Penguin | It doesn't care about extension 0 or *. It must be extenstion o. |
| 18:14.24 | jmetro | oh |
| 18:14.26 | jmetro | o as in "operator" |
| 18:14.30 | Penguin | Correct. |
| 18:14.33 | jmetro | i was thinking 0 as in 0perator |
| 18:14.43 | Penguin | Zero perator? |
| 18:14.53 | jmetro | 0 means operator to me ;) |
| 18:15.00 | j4jackj | On the phone keypad you gumby |
| 18:15.08 | Penguin | It means zero to everyone else, including asterisk. |
| 18:15.28 | j4jackj | GUMBY |
| 18:15.37 | Penguin | I'm Gumby, dammit! |
| 18:15.52 | jmetro | http://pastebin.com/1FWYUD7s |
| 18:15.56 | jmetro | that is my dialplan for my VM context |
| 18:16.00 | jmetro | still doesnt work even with subbing 0 for o |
| 18:16.28 | jmetro | or o for 0, whatever. |
| 18:17.14 | Katty | hi kids. |
| 18:17.18 | jmetro | ello guvnah |
| 18:17.23 | Penguin | Did you reload the voice mail module and dial plan after changing them? |
| 18:17.39 | jmetro | yes, trying again though |
| 18:18.10 | Penguin | You might either have to set the exitcontext to VM or put the o extension in the other context that led you to this one. |
| 18:18.15 | Katty | jmetro: WHAT DID YOU DO |
| 18:18.20 | jmetro | D= |
| 18:18.25 | Katty | hugs jmetro |
| 18:18.30 | jmetro | =D |
| 18:18.33 | Katty | jmetro: stoppppp breaking stuff! |
| 18:19.04 | Penguin | The verbose core output should reveal where the problem is. |
| 18:19.19 | jmetro | ive been watching console and it doesnt say anything though |
| 18:19.34 | Penguin | core set verbose 3 (or higher)? |
| 18:19.51 | jmetro | my verbose and debug lvels are set to 9001 each |
| 18:20.04 | Penguin | That's ridiculously excessive. |
| 18:20.07 | jmetro | Katty: I dont break things, i invent new ways of things not working. |
| 18:20.29 | Katty | jmetro: oh..well in that case... |
| 18:20.44 | Penguin | If there's no output, that makes me feel like the call isn't even getting to asterisk. |
| 18:20.55 | jmetro | by no output i meant no erros 8-| |
| 18:21.09 | Penguin | That's a lot different. |
| 18:21.12 | jmetro | i get to Voicemail(113@context) and from there, mash 0/* while listening to my unavailable message. |
| 18:21.35 | Katty | do you have that one thing enabled in voicemail.conf |
| 18:21.40 | jmetro | yus |
| 18:21.50 | Katty | and do you have the a and o context |
| 18:21.52 | Katty | err |
| 18:21.57 | Katty | it's not a context, it's in the context |
| 18:22.02 | Katty | oh what's it called? |
| 18:22.04 | jmetro | http://pastebin.com/VQQqYp8a my voicemail |
| 18:22.06 | Katty | digs through extensions.conf |
| 18:22.07 | Penguin | an extension. |
| 18:22.13 | jmetro | http://pastebin.com/1FWYUD7s my vm context |
| 18:22.23 | Penguin | Those things within a context are called extensions. |
| 18:22.25 | jmetro | its "o" and "*" |
| 18:22.32 | Katty | jmetro: here i'll pastebin for you dear |
| 18:23.00 | jmetro | i might sacrifice a hand to the elder gods if i wind up having some random typo |
| 18:23.10 | Penguin | If you press * during the outgoing vm message, it should exit to extension a. |
| 18:23.45 | Katty | jmetro: http://pastebin.ca/2449665 <- that may, or may not be useful |
| 18:23.46 | Penguin | Nevertheless, that would have appeared in core verbose output saying that you have no extension 'a'. |
| 18:24.27 | jmetro | no errors saying "no extension found", i have a, 0, o, and * |
| 18:24.35 | Penguin | Then something else is wrong. |
| 18:24.37 | Katty | * isn't... |
| 18:24.44 | Katty | * goes to uhh |
| 18:24.56 | Katty | words are failing me today |
| 18:25.06 | Katty | * goes to a |
| 18:25.13 | Penguin | That's what I already said. |
| 18:25.19 | jmetro | ooh you use isymphony too, waiting on isymphony 3 ? |
| 18:25.44 | Katty | i sur-pose. |
| 18:26.00 | Katty | that's just a little vm instance i use for testing stuff before i break things |
| 18:26.13 | jmetro | with any luck it will be finally multi-tenant so i dont have to put GUI clients on their own VM. |
| 18:26.30 | Katty | yay |
| 18:26.44 | jmetro | because showing all parks 100% of the time made sense, right |
| 18:27.15 | Katty | let's blame canada. |
| 18:27.28 | jmetro | with their beady little eyes , their trashcan mouths are full of lies. |
| 18:27.38 | Katty | it's the poutine. |
| 18:29.05 | Katty | jmetro: did you fix your exten => *,1,foo to exten => a,1,foo? |
| 18:29.15 | jmetro | ill try that |
| 18:30.13 | Katty | i like my exten with a little a,1,sauce |
| 18:30.34 | jmetro | <PROTECTED> |
| 18:30.36 | jmetro | then nothing |
| 18:30.38 | jmetro | =( |
| 18:30.48 | jmetro | i wonder if its the operator thing in vm.conf |
| 18:30.49 | Katty | that's that thing you have to enable in voicemail.conf |
| 18:30.52 | Katty | digs |
| 18:31.06 | jmetro | that was in my pb though, mailbox 113 has operator=yes |
| 18:31.14 | Penguin | During the voicemail playback, you pressed * and it didn't jump to extension a, right? |
| 18:31.21 | Katty | operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to <- blahblahblah |
| 18:31.23 | jmetro | i tried 0 and * |
| 18:31.42 | Katty | if you didn't enable operator=yes you're gonna have a bad time! |
| 18:31.44 | Penguin | operator is a per-vmbox option? |
| 18:31.46 | Katty | should start making asterisk memes |
| 18:32.26 | jmetro | i did voicemail show users for Context and it said 113 had operator on, i'll see if i can make it global |
| 18:32.40 | Katty | weird. i thought it was a global option in voicemail.conf |
| 18:32.45 | Katty | i mean, i only have the one enabled. |
| 18:33.21 | Penguin | I've never put it in a mailbox option list. |
| 18:33.32 | jmetro | tried [general] as well as the individual mailbox, no go |
| 18:34.26 | j4jackj | Should I make a PC speaker version of the 'Robot Dity' that Asterisk includes? |
| 18:34.33 | Katty | put it in [default] real quick instead of [context] |
| 18:34.43 | Katty | and then change your extensions conf to dial the mailbox@default |
| 18:34.54 | Katty | wait |
| 18:34.57 | Katty | waitwait. i remember this problem |
| 18:35.01 | Katty | it defaults to the context your vm box is in |
| 18:35.09 | Katty | so it's looking for exten => a in [Context] |
| 18:35.10 | jmetro | Oh. |
| 18:35.13 | jmetro | Oh. oh oh |
| 18:35.16 | Penguin | I said that. |
| 18:35.20 | jmetro | wait, wouldnt i still see the error? |
| 18:35.32 | Katty | i don't remember |
| 18:35.50 | Penguin | But we decided that the verbose output would have shown the failure. |
| 18:36.01 | Penguin | So that isn't likely to be the problem. |
| 18:36.10 | Katty | well he can try it any, mister penguin pantaloons. |
| 18:36.22 | Katty | s/any/anyway |
| 18:36.45 | Penguin | You could set the exitcontext and see if that leads to success. |
| 18:37.02 | jmetro | Hah |
| 18:37.04 | jmetro | It doesnt show in verbose. |
| 18:37.08 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 18:37.09 | jmetro | Katty =1 |
| 18:37.11 | jmetro | +1* |
| 18:37.20 | Katty | do i get a shiny blue ribbon for today? |
| 18:37.20 | jmetro | [Katty now has 350 concurrent calls!] |
| 18:37.27 | Katty | NO! |
| 18:37.39 | Katty | enables dnd |
| 18:37.47 | jmetro | Concurrent Calls is asterisk karma :3 |
| 18:37.56 | Katty | do not want concurrent calls :< |
| 18:38.02 | Katty | DO NOT WANT |
| 18:38.06 | jmetro | Lol |
| 18:38.22 | Katty | anyway, back to knitting. |
| 18:38.28 | jmetro | +1 for penguin too because he said it earlier but i forgot to check in my frenzy of 0 vs o |
| 18:38.50 | Katty | infobot: crittercam |
| 18:38.50 | infobot | from memory, crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 18:40.03 | jmetro | Did you knit that scarecrow shirt? |
| 18:40.57 | Katty | no i bought that at Goodwill. |
| 18:41.15 | Katty | do you lke how i defeated the purpose of putting up a scarecrow, but putting bird feeders on him? |
| 18:42.07 | jmetro | Its a feedcrow Domesticator, so you can have pet birds. |
| 18:42.13 | Katty | ^_^ |
| 18:43.36 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-68-168-116.home.otenet.gr) |
| 18:45.24 | Katty | drmessano: ping |
| 18:45.32 | Katty | drmessano: it's friday, yo. where you eat |
| 18:45.38 | Katty | drmessano: eat? at. |
| 18:46.34 | jmetro | hm |
| 18:46.53 | jmetro | set(CALLERID(NAME)=etc) right |
| 18:46.55 | Katty | what did you break now. |
| 18:46.58 | jmetro | D= |
| 18:47.03 | Katty | <3 |
| 18:47.06 | jmetro | =D |
| 18:47.13 | Katty | idk, i just use the number |
| 18:47.39 | Katty | infobot: thebook |
| 18:47.39 | 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 |
| 18:49.00 | Penguin | Set(CALLERID(name)=Your Name) |
| 18:49.17 | Penguin | Set(CALLERID(num)=1112223333) |
| 18:49.35 | Penguin | or |
| 18:49.40 | Penguin | Set(CALLERID(all)=Your Name <1112223333>) |
| 18:49.45 | jmetro | right |
| 18:49.59 | jmetro | lowercase name |
| 18:50.26 | Penguin | It would /probably/ work if you capitalized it, but I prefer to follow the instructions exactly. |
| 18:50.56 | jmetro | ugh, problem is spaces as always. i need sleep. |
| 18:51.07 | Katty | sleep?! what's that?! |
| 18:51.15 | Penguin | Spaces? What's wrong with spaces? |
| 18:51.29 | Katty | they keep expanding faster and faster. |
| 18:51.40 | jmetro | Set(CALLERID(NAME) = Operator-${CALLERID(NAME)} |
| 18:51.54 | Penguin | I see. Yeah, that's wrong. |
| 18:52.02 | jmetro | fixed it |
| 18:52.19 | Katty | i should tinker with my ltitle vmbox |
| 18:52.28 | Katty | i've still yet to register one box to another one |
| 18:52.31 | jmetro | Get realtime working on it, if you havent done realtime before. |
| 18:52.38 | Penguin | I should have eaten lunch. I have a rumbly in my tumbly. |
| 18:52.38 | jmetro | its super fun |
| 18:52.47 | Katty | Penguin: go eat. |
| 18:53.03 | Katty | maybe i can con jmetro into helping me register box b onto box a |
| 18:53.08 | Penguin | It'll have to wait a little longer. Have to get the little one off the school bus in a few minutes. |
| 18:53.26 | Katty | miniture hooman brigade! |
| 18:53.27 | Penguin | But then... |
| 18:53.45 | Penguin | I'm going to have a sammich of some variety. |
| 18:54.08 | jmetro | Katty: hows asterisk send out a register? |
| 18:54.14 | Katty | well, i think you some stuff in sip.conf on one box |
| 18:54.15 | Penguin | Probably one with lots of greasy meats, such as salami, pepperoni, and pastrami. |
| 18:54.22 | Katty | and then on the other one you do register => foo,stuff |
| 18:54.35 | jmetro | wait, i think i have this setup already actually katty |
| 18:54.42 | Katty | probably. |
| 18:54.49 | Katty | i've never needed to make two boxes talk before |
| 18:54.52 | Katty | so i've just never done it |
| 18:55.17 | jmetro | we do it for our conferencing |
| 18:55.30 | Penguin | Does at least one of the boxes have a dynamic IP address? |
| 18:55.42 | Katty | mmmm ,no |
| 18:55.45 | Katty | both are static. |
| 18:55.55 | Penguin | Then you don't need to register. |
| 18:55.55 | Katty | tho one will be behind a vpn at some point, not that it matters |
| 18:56.22 | Katty | exports the vm |
| 18:56.33 | paule32 | how can i register an asterisk server on differnet ip? also ip: 1.2.3.4 user 1234 password: secret ? |
| 18:56.44 | *** join/#asterisk monsterco (~monsterco@64.231.101.21) |
| 18:56.47 | monsterco | [TK]D-Fender> - I will try that - where can I get stock firmware Polycom Soundpoint IP 335 |
| 18:56.52 | Penguin | paule32: I have no idea what that question even means. |
| 18:57.07 | monsterco | By they way I brought their phones to our office now so I can play with them on my own time |
| 18:57.18 | jmetro | monsterco: probably on the polycom website. |
| 18:58.29 | Penguin | Asterisk to Asterisk (SIP to SIP): http://pastebin.com/Ag7tknm2 |
| 18:59.11 | Penguin | If one has a dynamic IP address, you'll need to register it to the static one. |
| 18:59.12 | Katty | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291281 <- that's what i found. |
| 18:59.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:59.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:59.35 | [TK]D-Fender | monsterco: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 18:59.36 | paule32 | Penguin, i have register a telephone number by my provider with a password, that number, i have setup the router - a avm fritzbox, that router tells me data how to setup, but i could only read the router address, the user name and the password, that i had to set |
| 18:59.45 | [TK]D-Fender | monsterco: 4.0.4 is the latest SIP general release for it |
| 19:00.07 | Katty | dear vmware. kindly export faster. |
| 19:00.08 | Penguin | katty: Mine is easier to follow. |
| 19:00.16 | Katty | Penguin: meh |
| 19:00.25 | Katty | Penguin: i'd rather learn it, not just copy dialplan. |
| 19:00.35 | Penguin | I don't waste time explaining how things work. I just show you the sample configuration and let you do it. |
| 19:00.54 | jmetro | Katty: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html is what i found |
| 19:01.15 | Penguin | They're going to show you what I've already shown you. |
| 19:01.17 | Katty | jmetro: ohah |
| 19:01.31 | Katty | jmetro: ty kindly |
| 19:01.58 | Katty | jmetro: we have eleventy billion sip trunks coming into one server |
| 19:02.05 | jmetro | that sounds like a lot. |
| 19:02.06 | Katty | jmetro: so i'm going to see if i can forward some calls to another one |
| 19:02.14 | Katty | eleventy billion, 2.... |
| 19:02.16 | Katty | all the same number |
| 19:03.03 | jmetro | I've heard counting is cyclical, but was never sure. |
| 19:03.47 | Katty | vsphere clearly can't count either. |
| 19:03.50 | Katty | it now says 8 minutes |
| 19:07.37 | Katty | yay! |
| 19:07.42 | Katty | deploys |
| 19:08.51 | j4jackj | 1111111, 2, 3333, 4,52455252 |
| 19:08.59 | jmetro | hm |
| 19:09.28 | jmetro | i have two duplicate contexts, one sets the callerid to OPERATOR-(cid name) other sets it to CALLGROUP-(cid name) |
| 19:09.58 | jmetro | perfectly the same aside from that, but the CALLGROUP one wont show up as CALLGROUP-(cidname) on their phone =( |
| 19:10.27 | j4jackj | ;can you pastebin them? |
| 19:10.54 | j4jackj | hi pabelanger |
| 19:10.58 | Katty | pastebins jmetro |
| 19:11.12 | j4jackj | Katty: no time for silliness. |
| 19:11.19 | jmetro | is pasted into a bin. oh lawd @@ |
| 19:11.21 | Katty | i'm all srs business. |
| 19:11.32 | paule32 | register => fromuser@fromdomain:secret@host |
| 19:12.11 | j4jackj | jmetro: can you pastebin the two contexts? |
| 19:12.16 | paule32 | is fromuser the extension? fromdomain the server ip from asterisk? and host the provider? |
| 19:12.58 | jmetro | http://pastebin.com/RNzBtXcJ |
| 19:13.02 | j4jackj | paule32: no. fromuser:secret@hostname/extension. fromuser is the username on the server. |
| 19:13.11 | jmetro | whats funny is my debug verboses are actually showing CALLGROUP(cid) |
| 19:13.20 | jmetro | but the phone / op panel doesnt |
| 19:14.13 | paule32 | j4jackj, is /extension requiered? |
| 19:14.27 | jmetro | Oh. Nevermind. Now it does. |
| 19:14.28 | j4jackj | paule32: no. |
| 19:14.39 | jmetro | sigh. Im a huge waste of time <.< |
| 19:14.46 | paule32 | ok thank you |
| 19:15.12 | paule32 | j4jackj, how can i check if the server registered |
| 19:15.18 | paule32 | in the console |
| 19:15.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.16) |
| 19:16.38 | j4jackj | paule32: It will say something to that effect if you launched with high verbosity |
| 19:17.56 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 19:19.24 | jeffspeff | i have a shell script that i need to run. i have 1 variable to pass the script from the dialplan and i expect 1 variable back from the script to use the in the dialpplan. from what i've read only System() lest you send args to the script and only SHELL() lets you retrieve results from the script. is there any way to send args to the script and get a result back? |
| 19:19.46 | j4jackj | Owhatta? |
| 19:19.48 | jmetro | jeffspeff: i have a solution |
| 19:20.11 | jeffspeff | yes? |
| 19:21.15 | jmetro | same => n,Set(var=${SHELL(php /myscript.php ${myarg})}) |
| 19:21.47 | jeffspeff | the documentation for SHELL() doesn't say that it is able to pass any args |
| 19:22.01 | jmetro | php does it. |
| 19:22.05 | jmetro | php script args |
| 19:22.10 | jmetro | passes args to phpscript |
| 19:22.46 | karl-s | jmetro, I dont have that function... core show function SHELL returns nothing. What version do you get that function from? |
| 19:22.48 | jeffspeff | hmm... didn't know that |
| 19:23.28 | monsterco | [TK]D-Fender> - so I see a 5.0.0 combined and split release for polycom firmware - I guess I will grab the combined one |
| 19:23.55 | jmetro | Connected to Asterisk SVN-branch-11-r378219 |
| 19:24.00 | paule32 | cool "sip show registry" shows 1 sip registration |
| 19:24.01 | jmetro | shell works on everything though |
| 19:24.04 | monsterco | 427 MB - holy cow - something wrong lol |
| 19:24.05 | *** join/#asterisk amessina_ (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:24.10 | paule32 | super thing this pbx |
| 19:24.39 | Penguin | paule32: Then you have registered your asterisk to one other peer. |
| 19:25.04 | paule32 | Penguin, and i can now do outgoing calls? |
| 19:25.06 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:25.08 | Penguin | No. |
| 19:25.12 | monsterco | actualy 187MB for 4.0.4 combined |
| 19:25.24 | Penguin | The fact that you registered to another peer has nothing to do with if you can make outgoing calls. |
| 19:25.30 | monsterco | that seems like a lot for a phone firmware |
| 19:25.31 | Penguin | It just means you have registered. |
| 19:25.47 | paule32 | hmm ok |
| 19:26.00 | Penguin | To send calls, you must have dial plan. |
| 19:26.00 | paule32 | have to do setup extension? |
| 19:26.03 | Penguin | Yes. |
| 19:26.10 | Penguin | You have to have an extension that calls something. |
| 19:26.13 | [TK]D-Fender | [15:23]monsterco[TK]D-Fender> - so I see a 5.0.0 combined and split release for polycom firmware - I guess I will grab the combined one <- 5.0 = useless |
| 19:27.08 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:27.35 | drmessano | Katty, sup foo |
| 19:29.24 | jmetro | karl-s: coincidentally i could not find SHELL on mine either, but it works. |
| 19:29.33 | jmetro | karl-s: i spent a long while getting that part working because I couldnt find shell |
| 19:34.55 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 19:35.43 | gmalsack | hey guys ~ ast 11.5 is converting the hash in my dial command to %23. anyone know how to stop that? |
| 19:35.44 | monsterco | seems like anything above 4.0.4 is not supported by IP 335 |
| 19:36.27 | Penguin | core show function SHELL |
| 19:38.34 | monsterco | [TK]D-Fender> - what do you mean useless though? |
| 19:39.19 | [TK]D-Fender | monsterco: See that "no" in the IP 335 column? It means "no" |
| 19:40.04 | *** join/#asterisk dorphalsig (~dorphalsi@181.50.255.162) |
| 19:40.11 | dorphalsig | Hi |
| 19:40.31 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 19:41.31 | monsterco | [TK]D-Fender> - that is why I said I can't use 5.0.0 - and said that I will use 4.0.4 - but you said anything less than 5.0.0 is useless |
| 19:41.51 | monsterco | " <- 5.0 = useless" |
| 19:42.13 | [TK]D-Fender | monsterco: that was an ARROR |
| 19:42.16 | [TK]D-Fender | ARROW* |
| 19:42.21 | [TK]D-Fender | not "less than" |
| 19:43.50 | dorphalsig | I'm running freepbx 2.11 and asterisk 11, and I'm having this strange issue when a caller goes into a queue. Even if there are available agents in the queue, teh call just lurks around and I'm not seeing the state of the devices change to ringing |
| 19:44.08 | dorphalsig | I dunno if this is a freepbx issue or an asterisk one |
| 19:44.11 | monsterco | [TK]D-Fender>- great thanks |
| 19:44.12 | jmetro | i would check in #freepbx as its hard for us to troubleshoot their custom config files. |
| 19:44.44 | [TK]D-Fender | monsterco: I use it to separate my response from the text I copy from you that I am responding to |
| 19:45.11 | monsterco | [TK]D-Fender> - speaking of everything messed up with Polycom - it took me 15 minutes to download the firmware - their servers are SLOW but it's with me now and it's with me now - so I guess I will go to boot room and point the phone to my pc with ftp server? |
| 19:45.22 | monsterco | yep got it |
| 19:47.10 | dorphalsig | jmetro2026 the queue files are not very complex. can you give it a look? If it doesnt look right I'll just try and find somebody in freepbx to help me out |
| 19:47.11 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:47.49 | [TK]D-Fender | monsterco: get the a newer bootrom image as well. |
| 19:47.58 | [TK]D-Fender | monsterco: go through their product page to get that |
| 19:48.09 | [TK]D-Fender | monsterco: bootrom + sip app are 2 differnt parts |
| 19:48.31 | monsterco | oh lala - speak of complications |
| 19:50.09 | monsterco | it just can't be anymore confusing - http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip335.html |
| 19:50.21 | monsterco | look at that ^^^^ why so many different products listed on on epage |
| 19:50.27 | monsterco | *one page* |
| 19:51.20 | monsterco | [TK]D-Fender - this is what I got from URL you referenced: Polycom_UC_Software_4_0_4_release_sig_combined |
| 19:51.23 | monsterco | it's a zip file |
| 19:52.30 | monsterco | [TK]D-Fender - for boot room I see this: http://downloads.polycom.com/voice/voip/uc/spip_ssip_BootROM_4_3_1_release_sig.zip |
| 19:52.35 | monsterco | am I on the right track? |
| 19:53.37 | [TK]D-Fender | looks about right |
| 19:54.32 | monsterco | and you said I need another file too? a SIP firmware? |
| 19:54.58 | monsterco | oh I see that is the SIP one |
| 19:55.21 | monsterco | so I am putting both on HTTP server and will point phone to it - any other config files I have to put there? and do I have to unzip these? |
| 19:57.47 | [TK]D-Fender | monsterco: I recommend FTP |
| 19:58.09 | paule32 | [Sep 13 21:57:11] WARNING[23356]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 19:58.09 | paule32 | [Sep 13 21:57:11] WARNING[23356]: app_dial.c:2385 dial_exec_full: Invalid timeout specified: '$(EXTEN-1)'. Setting timeout to infinite |
| 19:58.25 | [TK]D-Fender | if you extract thos as-is and point it there for provisioning it should completely wipe the phone and update to the latest |
| 19:58.41 | monsterco | so do unzip them |
| 19:59.07 | [TK]D-Fender | paule32: I already explained the first (device not registered), the 2nd is an invalid variable reference because of your improper () |
| 19:59.11 | [TK]D-Fender | monsterco: yes |
| 19:59.16 | [TK]D-Fender | monsterco: all in same folder |
| 19:59.29 | paule32 | http://codepad.org/icWhd9sl |
| 20:00.48 | [TK]D-Fender | paule32: exten => _0X.,1,Dial(SIP/deutschland,$(EXTEN-1)) <- you are using a comma improperly there... the number is not a 2nd parameter |
| 20:00.54 | [TK]D-Fender | pauland $90 is wrong |
| 20:00.58 | [TK]D-Fender | $() |
| 20:03.00 | jmetro | dfender just cant type today. |
| 20:03.23 | [TK]D-Fender | nope |
| 20:03.35 | [TK]D-Fender | my back is killing me.... |
| 20:03.40 | [TK]D-Fender | can't wait to get out of here...\ |
| 20:05.23 | monsterco | I unzipped it all and phone says: uploading log file. - what the heck does that mean |
| 20:05.24 | jmetro | whats wrong with yours? Mine's got a lumbar strain |
| 20:05.30 | monsterco | I thought there are no logs on GUI |
| 20:05.48 | [TK]D-Fender | jmetro: Muscle re-adjustment following a broken collar-bone |
| 20:06.13 | [TK]D-Fender | monsterco: Thre is also a BOOT LOG so you can see how it is acting live |
| 20:06.21 | [TK]D-Fender | monsterco: it's for more than just configs |
| 20:07.27 | monsterco | how is that accessed? |
| 20:08.00 | jmetro | [TK]D-Fender: well that sounds awful compared to chronic lower back pain [hopefulyl temporary] |
| 20:08.18 | [TK]D-Fender | monsterco: It's a text file .. on an FTP server ... use your imagination :) |
| 20:08.24 | monsterco | downloading bootROM please wait |
| 20:08.32 | Penguin | Chronic and temporary? |
| 20:08.58 | [TK]D-Fender | monsterco: go grab a coffe... it'll be 5-10 |
| 20:09.18 | monsterco | looks like almost done for bootROM - 2 mins from my server - 15 from polycom |
| 20:09.22 | monsterco | rebooting it says now |
| 20:09.28 | monsterco | formatting file system plz wait |
| 20:09.32 | [TK]D-Fender | there will be another 2 reboots most likely |
| 20:09.32 | monsterco | rebooting again |
| 20:09.55 | monsterco | waiting for network - it wait's like 30 seconds to get ip - pretty lame |
| 20:10.01 | monsterco | updating sip.id please wait |
| 20:11.26 | jmetro | man, we dont need a blow by blow. Just let your phone come up |
| 20:12.41 | paule32 | [Sep 13 22:12:01] WARNING[24088]: app_dial.c:2127 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 20:13.13 | [TK]D-Fender | paule32: I highly recommend paying close attention to your syntax... |
| 20:13.18 | Penguin | That's a little different from what I remember it saying. I don't recall that colon. |
| 20:13.33 | gmalsack | so no one knows how to stop *11.5 from converting the hash symbol in my dial command to %23??????? |
| 20:13.52 | [TK]D-Fender | gmalsack: Show us the complete call with SIP debug |
| 20:14.05 | gmalsack | pastebin coming... |
| 20:14.21 | Gugge | gmalsack: why would it convert hash to "%23???????" |
| 20:14.51 | jmetro | gmalsack: i cant imagine why that would happen, what text editor are you using. |
| 20:15.19 | gmalsack | pastebin.com/4CUeNyAr |
| 20:15.35 | karl-s | %23 is Unicode for Hash '#' |
| 20:15.45 | jmetro | im thinking its his text editor |
| 20:15.55 | Penguin | I think you're on the right track. |
| 20:16.07 | gmalsack | asterisk console... sip debug.... not an editor |
| 20:17.14 | gmalsack | TK: thoughts??? |
| 20:17.16 | Qwell | <rant>Who the heck decided that naming another Android phone the G2 was a good idea? Stupid LG.</rant> |
| 20:17.33 | Penguin | If it is just an appearance problem in your console, adjust your terminal settings. |
| 20:17.45 | [TK]D-Fender | gmalsack: pedantic=no <- under [general]\ |
| 20:17.52 | gmalsack | thanks.... |
| 20:18.15 | jmetro | i find this ham to be shallow and pedantic |
| 20:19.29 | drmessano | Qwell, did they run out of verbs to name Android phones already? |
| 20:19.36 | gmalsack | TK: nope... |
| 20:19.48 | drmessano | I was quite looking forward to the Samsung Thwap |
| 20:19.52 | Qwell | drmessano: No, the next HTC phone will be the Tickler. |
| 20:20.43 | Qwell | drmessano: ( http://techcrunch.com/2012/03/26/condom-or-android-handset-name/ ) |
| 20:21.28 | gmalsack | TK: nevermind. forgot that I was also playing with the dialstring. changed that back and now it's working... thanks again! |
| 20:21.42 | jmetro | I'd rather be talking on the Samsung Magnum XL than have apple be building a database of my fingerprints to hand over to the government @.@ |
| 20:22.12 | drmessano | Yeah, because Apple introduced the first device with a fingerprint scanner, ever |
| 20:22.17 | gmalsack | just got the new motorola mini..... awesome phone! wouldn't have minded the others, just don't like the bigger screens. |
| 20:22.25 | Gugge | jmetro: dont worry about the fingerprint stuff, they have your print already :P |
| 20:22.36 | jmetro | Apple introduced the idea that nothing on your personal electronics is private and they have the right to pull your GPS data every 5 minutes |
| 20:22.43 | gmalsack | gugge: LMFAO!!! |
| 20:22.43 | drmessano | jmetro, you're probably already in a database anyway. |
| 20:22.46 | jmetro | fingerprint data would be a lot smaller. |
| 20:23.03 | paule32 | is it possible to combine context's ? |
| 20:23.09 | jmetro | #include |
| 20:23.11 | Qwell | paule32: explain what you mean |
| 20:23.11 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:23.21 | Penguin | Context's what? |
| 20:23.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.16) |
| 20:24.00 | paule32 | context local with user 1001 and context country with user 1001 |
| 20:24.58 | [TK]D-Fender | paule32: go read up on your dialplan basics.... |
| 20:25.00 | [TK]D-Fender | ~book |
| 20:25.00 | 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 |
| 20:25.12 | [TK]D-Fender | paule32: Specifically on "include"'s |
| 20:25.28 | drmessano | At least the Pedophiles won't have to worry about fingerprint scanners, because they'll all be carrying Ticklers in the pocket of their clown suits running the lastest Android KitKat or Almond Joy |
| 20:25.33 | drmessano | drops mic |
| 20:25.52 | Penguin | You can include one context in another context, but with duplicated extensions, the one in the included context won't be used. |
| 20:26.07 | Qwell | drmessano: Don't be silly. Almond Joy doesn't start with an L. |
| 20:26.15 | Qwell | drmessano: aLmondjoy, obviously. |
| 20:26.24 | jmetro | Android Lemon Drop |
| 20:26.25 | drmessano | lol |
| 20:27.25 | drmessano | Qwell, $10 says M will be Mounds. Pervs and 12 yr olds will cheer in the streets after being given at least 2 years of "Mounds" jokes |
| 20:27.31 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:27.38 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 20:27.45 | paule32 | also some user's that have context "local" with a range of numbers and a group of users (may be the same) with global numbers |
| 20:28.09 | paule32 | i have to add all users in separate files, and then iclude them in contexts? |
| 20:28.10 | drmessano | Qwell, "Wanna see my Mounds.. HER HER HER" |
| 20:29.06 | drmessano | I think all Asterisk releases after 12 should use code names based on brands of cereal |
| 20:29.21 | drmessano | I can't wait to upgrade to Lucky Charms |
| 20:30.15 | Qwell | mjordan: ^^^^^^^^^^^^^^^^^^^^ |
| 20:30.25 | jmetro | paule32: you might want to read the book fully, as i dont think you understand your own question. |
| 20:30.27 | Qwell | also, 12 isn't released yet, so why wait? |
| 20:31.22 | drmessano | 12 would be Lucky Charms if we're associated 12 = L. That was a happy accident |
| 20:31.22 | Qwell | drmessano: "The Asterisk Development Team is proud to announce the release of Asterisk: Count Chocula." |
| 20:32.00 | drmessano | Qwell, I was thinking about [TK]D-Fender laying into someone because they didn't upgrade from Frankenberry to Golden Grahams |
| 20:32.20 | paule32 | jmetro, http://codepad.org/O7WyGXyW |
| 20:32.24 | jmetro | Apple Jacks |
| 20:32.28 | Qwell | You're still using Asterisk WITH Crunchberries? |
| 20:32.33 | drmessano | hahaha! |
| 20:32.43 | jmetro | What would B, be? |
| 20:32.54 | drmessano | We haven't supported Cinnamon Toast Crunch in 3 years |
| 20:32.59 | drmessano | Derp |
| 20:32.59 | _Corey_ | Sounds like you guys are on to something here... |
| 20:33.26 | jmetro | CTC and Crunchberries are both Ast3.. |
| 20:33.47 | jmetro | Apple Jacks |
| 20:33.50 | jmetro | Baron von Redberry |
| 20:33.54 | jmetro | Cinnamon Toast Crunch |
| 20:34.05 | jmetro | Dyno-Bites |
| 20:34.17 | Qwell | I'm vetoing Cinnamon Toast Crunch, for Cookie Crisp. |
| 20:35.19 | jmetro | I dont know any E's. |
| 20:35.29 | drmessano | What about Captain Crunch? Are we not paying tribute to John Draper, the Whistle, and the memory of the roof of our mouths there? |
| 20:35.51 | Qwell | http://en.wikipedia.org/wiki/List_of_breakfast_cereals |
| 20:36.05 | drmessano | wonders how much skin from the roof of his mouth he consumed as a child. SCRAAAAAPE |
| 20:36.08 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 20:36.19 | jmetro | Cap'n is saved for Peanut Butter Capn Crunch, being the superior flavor. |
| 20:37.06 | drmessano | and no healthy cereals.. |
| 20:37.17 | drmessano | Highest sugar and fat content FIRST |
| 20:37.29 | drmessano | Most importantly the sugar |
| 20:38.02 | drmessano | The lack of E cereals is disheartening |
| 20:38.23 | monsterco | [TK]D-Fender> - its still "Updating sip.id please wait." - I guess it takes longer than installing Windows ME |
| 20:39.35 | drmessano | monsterco, the BSOD and reboot after Windows Me install technically did not count as part of the install. It was more like an intermission |
| 20:39.50 | drmessano | A good excuse to go outside and throw a frisbee.. or a brick |
| 20:45.50 | mjordan | I'm down with FrankenBerry |
| 20:49.49 | monsterco | drmessano - it takes install time and BSOD time and ........... |
| 20:50.14 | monsterco | polycom engineers are the worst i realize |
| 20:50.45 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:51.21 | *** join/#asterisk justincampbell (~justincam@pool-173-61-28-102.cmdnnj.fios.verizon.net) |
| 20:52.05 | paule32 | how can i show the channeltypes in cli? |
| 20:52.26 | monsterco | sip show channels |
| 20:52.40 | justincampbell | i had to patch chan_sip.c for a client of mine, because a peer (Avaya CS1K) is sending multiple Authorization headers (sometimes 10 or 20 of them) and the first one has a stale nonce |
| 20:52.44 | justincampbell | https://github.com/justincampbell/asterisk-1.8-current/pull/1/files |
| 20:53.26 | justincampbell | so i was wondering, is it worth cleaning this up and getting it contributed back to the asterisk project? is there a better way to fix the issue? |
| 20:55.47 | [TK]D-Fender | checkout time, BBIAB |
| 20:57.18 | BKhan | Hi. I need help regarding to DTMF input. When we take DTMF input asterisk 10.5.1 accept from hard phone but when we give it from softphone it do not accept |
| 20:57.51 | jmetro | justincampbell: it sounds like you should update to 11 |
| 20:58.12 | justincampbell | jmetro: ok thanks, ill suggest that to the client |
| 20:58.16 | jmetro | BKhan: make sure your softphone is set to the proper dtmf setting, like rf2883 or whatever it is |
| 20:58.38 | jmetro | justincampbell: that might help, if its still an issue after the update, it would probably help a lot |
| 20:58.43 | justincampbell | are there breaking changes with AMI/AGI? |
| 20:58.55 | justincampbell | jmetro: ^^ 1.8 to 11 ? |
| 20:59.28 | jmetro | I dont use either so im not sure. |
| 20:59.40 | justincampbell | we have a ton of application logic written around AMI/AGI and the psql database format |
| 20:59.51 | jmetro | there are notes about it if you look them up |
| 20:59.55 | justincampbell | and they don't have full-time developers anymore, they just pay me to do maintenence :? |
| 21:00.03 | justincampbell | ok thanks jmetro, ill take a look at the changelog |
| 21:00.15 | BKhan | How can we check DTMF setting from soft phone |
| 21:00.23 | BKhan | I am using EyeBEam |
| 21:01.16 | jmetro | BKhan: i have no idea..on my softphones i just go to settings and click everything thats clickable until i find the option im looking for. |
| 21:01.33 | karl-s | I would also check with the softphone manual |
| 21:01.53 | karl-s | Theres too many softphone to memorize every setting for each |
| 21:02.51 | BKhan | jmetro: normally i also not change any settingand DTMF works perfectly |
| 21:02.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.16) |
| 21:03.35 | jmetro | BKhan: i'm not the IT for Eyebeam, im offering what sounds like the problem. |
| 21:04.49 | monsterco | says wrong image |
| 21:04.51 | monsterco | which one is that |
| 21:05.37 | BKhan | jmetro: is it may be issue of version? Actually i am using it as bridge server and i am using confbridge as bridge sever when we give pin number asterisk do not accept |
| 21:06.03 | jmetro | BKhan: simple test - call an IVR and see if it does dtmf |
| 21:06.58 | paule32 | is the peer name the channel name? |
| 21:07.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.16) |
| 21:14.40 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:16.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:16.20 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 21:17.54 | navaismo | ~book |
| 21:17.55 | 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 |
| 21:17.57 | navaismo | 4book |
| 21:18.23 | navaismo | awesome the bot avoid spam |
| 21:18.29 | jmetro | indeed |
| 21:19.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.16) |
| 21:21.58 | monsterco | I am getting wrong bootROM image - hmmmm - which one is the right one? |
| 21:21.59 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 21:23.37 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 21:24.08 | monsterco | [TK]D-Fender - I am confused when I look at this page but it seems it has all the needed stuff - can you please point out which files I need? http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip321_331.html |
| 21:24.16 | monsterco | Polycom Soundpoint IP 335 |
| 21:24.29 | [TK]D-Fender | SIP & BR |
| 21:24.44 | [TK]D-Fender | of course it's good to have the admin guides to match |
| 21:24.49 | monsterco | I downloaded this but it tells me wrong bootROM: spip_ssip_BootROM_4_3_1_release_sig.zip |
| 21:26.07 | monsterco | [TK]D-Fender - there is UC, VVX, ............ |
| 21:26.11 | [TK]D-Fender | What tells you that/ |
| 21:27.38 | monsterco | the web page i referenced above |
| 21:27.55 | monsterco | [TK]D-Fender - http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip321_331.html |
| 21:29.04 | [TK]D-Fender | monsterco: I still am not clear on what you have and what you're comparing it to |
| 21:29.19 | [TK]D-Fender | monsterco: And where we left off was my only knowing about your having IP 335's |
| 21:36.48 | *** join/#asterisk mohadib (~mohadib@unaffiliated/mohadib) |
| 21:45.09 | monsterco | [TK]D-Fender> - the phone says I have the wrong bootROM image. can you point me to right bootROM image link |
| 21:45.09 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 21:46.00 | monsterco | I am confused when I look at Polycoms page here: http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip335.html - there are lots of versions of BootROM and SIP - can you point out specifically which one is right for IP 335? |
| 21:54.20 | monsterco | [TK]D-Fender> - ^^^^ hanks |
| 21:54.22 | monsterco | thanks* |
| 21:56.57 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 22:00.55 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 22:05.40 | monsterco | Anyone here knows which version of bootROM firmware works with Polycom IP 335? |
| 22:07.35 | carrar | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 22:08.11 | carrar | oh thats close |
| 22:10.37 | *** join/#asterisk paule32 (~paul@dslb-188-106-252-082.pools.arcor-ip.net) |
| 22:10.47 | monsterco | carrar - that's the UC and not the bootROM |
| 22:10.55 | carrar | What do you want to run? |
| 22:11.17 | carrar | that phone supports up to bootrom 4.3.1 |
| 22:11.31 | carrar | depending the sip application you run |
| 22:12.24 | monsterco | I want the latest Release - is that 4.3.1? |
| 22:12.43 | monsterco | carrar - ^^^ |
| 22:12.44 | carrar | so 3.2.7 SIP code is what you are running? |
| 22:12.52 | monsterco | I don't know what I am running |
| 22:12.58 | carrar | I think thats the l ast before it jumps to UC |
| 22:13.07 | monsterco | this phone is now in mode to download software - I don't see about version |
| 22:13.24 | monsterco | ok - so I can't jump versions in btw |
| 22:13.24 | carrar | http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip335.html |
| 22:13.36 | carrar | You may want to read up befor you attempt to change firmware |
| 22:13.46 | monsterco | carrar - and again many versions posted there which confuses me |
| 22:13.52 | carrar | yes |
| 22:14.00 | carrar | So if you do not want UC |
| 22:14.08 | monsterco | read in which guide? there are many guides too - and release notes for 4.3.1 says it's good for IP 335 |
| 22:14.23 | monsterco | I just want the phone to connect to Asterisk |
| 22:14.24 | carrar | then 3.2.7 SIP split is the sip app that I would use |
| 22:14.32 | carrar | and the bootrom would be, looking that up |
| 22:14.45 | [TK]D-Fender | 4.04 as I told originally |
| 22:15.20 | monsterco | carrar - you mean this one: http://downloads.polycom.com/voice/voip/sp_ss_sip/SoundPoint_IP_SIP_3_2_7_release_sig_split.zip |
| 22:15.21 | monsterco | ? |
| 22:15.31 | [TK]D-Fender | that is SIP, not the bootrom |
| 22:15.32 | monsterco | that is the split and what about the sip bootROM? |
| 22:15.34 | [TK]D-Fender | do NOT mix those up |
| 22:15.43 | [TK]D-Fender | SIP is SIP |
| 22:15.43 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ygbcmlriridwzopi) |
| 22:15.46 | [TK]D-Fender | BootROM is BootROM |
| 22:15.51 | [TK]D-Fender | TWO parts |
| 22:15.56 | monsterco | [TK]D-Fender> - I have the 4.0.4 and I don't think that is the problem |
| 22:16.18 | monsterco | I also have "spip_ssip_BootROM_4_3_1_release_sig.zip" and that seems to be the problem |
| 22:16.30 | monsterco | so which version should I downgrade the "spip" to? |
| 22:16.41 | carrar | *sigh* |
| 22:16.46 | carrar | You need to stop |
| 22:16.50 | carrar | and read the admin guide |
| 22:17.06 | carrar | actually |
| 22:17.28 | carrar | I don't see that data in there |
| 22:17.32 | carrar | I forgot where that is |
| 22:18.32 | monsterco | lol - it's because polycom website is fucked up |
| 22:18.41 | monsterco | i read their release notes and it says it works |
| 22:18.49 | monsterco | for 335 - i am going down to 4.2.1 now |
| 22:18.58 | carrar | http://downloads.polycom.com/voice/voip/relnotes/SIP_Software_Release_Notes_3_2_7.pdf |
| 22:19.13 | carrar | in the release notes for the SIP code you want to run tells you what bootROM to use |
| 22:20.45 | carrar | SIP Application should work with BootROM 4.2.1 |
| 22:20.53 | carrar | SIP Application 3.2.7 that is should work with BootROM 4.2.1 |
| 22:21.11 | monsterco | [TK]D-Fender - in file name it says "sip" and "bootROM" |
| 22:21.20 | [TK]D-Fender | show me |
| 22:21.32 | monsterco | spip_ssip_BootROM_4_2_1_release_sig.zip |
| 22:21.32 | [TK]D-Fender | You are mashing this up pretty bad... |
| 22:21.37 | monsterco | lol yes |
| 22:21.40 | [TK]D-Fender | monsterco: NO\ |
| 22:21.42 | [TK]D-Fender | SPIP |
| 22:21.44 | [TK]D-Fender | SountPoint IP |
| 22:21.45 | monsterco | that is the file that downloaded |
| 22:21.47 | [TK]D-Fender | not ***SIP*** |
| 22:21.48 | carrar | Thats bootROM 3.2.1 |
| 22:21.54 | carrar | 4.2.1 |
| 22:22.01 | monsterco | what about "ssip" |
| 22:22.05 | carrar | what about it |
| 22:22.26 | [TK]D-Fender | [18:21]monstercospip_ssip_BootROM_4_2_1_release_sig.zip <- this is NOT the SIP APPLICATION |
| 22:22.31 | monsterco | carrar - so I need bootROM 4.21 or 3.2.1? |
| 22:22.38 | carrar | Soundpoint Soundstation IP |
| 22:22.50 | carrar | YOU REALLY SHOULD READ |
| 22:22.53 | carrar | the docs |
| 22:22.59 | monsterco | just doing that |
| 22:22.59 | *** part/#asterisk justincampbell (~justincam@pool-173-61-28-102.cmdnnj.fios.verizon.net) |
| 22:23.42 | carrar | looks at TK |
| 22:23.58 | *** join/#asterisk cian1500ww (~cian@unaffiliated/cian1500ww) |
| 22:24.11 | carrar | Whats on the actively list for the weekend |
| 22:24.46 | carrar | Polycom phones are great, but they do take a little time to learn |
| 22:24.59 | carrar | People get fustrated with them cause there are so many options |
| 22:25.06 | monsterco | can you link me to files I need or you guys not sure either? |
| 22:25.16 | carrar | http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip335.html |
| 22:25.28 | carrar | I am absolutely sure they are there |
| 22:25.44 | carrar | You're going to find everything you need and more tehre |
| 22:25.57 | carrar | Polycom is awesome that way |
| 22:26.01 | monsterco | carrar - lol - dude that page is confusing - 100s of versions and I already downloaded latest one that shows supported on the list and phone tell me no wrong firmware |
| 22:26.11 | carrar | If it's confusing go do something else |
| 22:26.11 | monsterco | i am going to start from lowest firmware |
| 22:26.16 | carrar | then come back |
| 22:26.25 | carrar | cause you need to jsut carefully read it |
| 22:26.49 | carrar | So |
| 22:27.01 | carrar | You already told me you do not want to run a UC version |
| 22:27.12 | carrar | so now you know what to look for as far as SIP Application code |
| 22:27.19 | carrar | the latest that isn't UC |
| 22:27.21 | paule32 | the VoipCarrier in sip.conf, is that the router where calls come from? |
| 22:27.30 | carrar | Right? |
| 22:27.53 | carrar | monsterco, stay focused here |
| 22:28.33 | carrar | ok |
| 22:28.36 | carrar | good lucky |
| 22:28.39 | carrar | luck |
| 22:29.02 | [TK]D-Fender | BootROM 4.3.1 is supported on the 335 |
| 22:30.44 | monsterco | carrar - lol kinda hard |
| 22:31.03 | monsterco | the version tk-d-fender told me is not working on this phone - maybe because of jump in versions |
| 22:31.05 | carrar | Get SIP Application 3.2.7 & bootROM 4.3.1 |
| 22:31.09 | monsterco | so I am using lowestone and go one by one |
| 22:31.46 | carrar | why |
| 22:32.21 | carrar | Why not just read the release notes for bootROM 4.3.1 first? |
| 22:33.04 | carrar | read section 1 |
| 22:33.37 | carbinemonoxide | I am having issues with asterisk connecting calls using Google Voice. When I call someone, if they pick up my SIP phone continues to ring. Asterisk sees that they pick up but never connects us. http://pastie.org/private/mo6xjd91rrfvo5aj6oytg |
| 22:34.43 | carbinemonoxide | It was working fine then one day, without me changing anything it started doing this. |
| 22:35.08 | monsterco | I did and it says supported - but phone says Wrong BootROM - what can I do other than lowering the version |
| 22:35.23 | monsterco | unless again this has something to do with the phone being locked to RingCentral |
| 22:35.35 | [TK]D-Fender | monsterco: Polycom_UC_Software_4_0_4_release_sig_split.zip |
| 22:35.41 | [TK]D-Fender | monsterco: Appears to contain the bootrom as well |
| 22:35.50 | [TK]D-Fender | monsterco: You should be able to use it alone, as-is |
| 22:36.17 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 22:36.35 | WIMPy | The three day battle with one phone. |
| 22:36.37 | monsterco | the lowest ones just booted the phonefine |
| 22:36.45 | monsterco | WIMPy - yes - hehehe |
| 22:36.47 | monsterco | well there are many |
| 22:36.51 | monsterco | one good - rest will be good |
| 22:36.59 | carrar | or just download from the list |
| 22:36.59 | carrar | http://downloads.polycom.com/voice/voip/uc/spip_ssip_BootROM_4_3_1_release_sig.zip |
| 22:38.39 | [TK]D-Fender | [18:35][TK]D-Fendermonsterco: Appears to contain the bootrom as well <- |
| 22:38.42 | [TK]D-Fender | Just do it |
| 22:40.11 | monsterco | freakin phone just started working |
| 22:40.32 | [TK]D-Fender | \o/ |
| 22:40.32 | monsterco | now time to test the latest that [TK]D-Fender> mentioned - can you directly link me [TK]D-Fender>? |
| 22:40.51 | [TK]D-Fender | I already di, multiple times |
| 22:40.58 | monsterco | no, url link |
| 22:41.00 | carrar | hahah |
| 22:41.08 | [TK]D-Fender | http://support.polycom.com/PolycomService/support/us/support/eula/ucs/UCagreement_4_0_4_split.html |
| 22:41.10 | monsterco | i don't want to guess again |
| 22:41.14 | [TK]D-Fender | Right off the matrix screen |
| 22:41.15 | carrar | Do need anyone to google for you too? |
| 22:41.18 | monsterco | perfect ;) |
| 22:41.19 | monsterco | thanks |
| 22:41.25 | monsterco | oh that would be amazing |
| 22:42.02 | WIMPy | I know someone who calls people to ask them to google something for him :-( |
| 22:42.14 | monsterco | lol |
| 22:42.21 | monsterco | there is a business like that |
| 22:42.32 | WIMPy | Well, The guy is teacher. Maybe that's an excuse. |
| 22:42.33 | monsterco | you call and ask the personel assistant whatever |
| 22:42.46 | monsterco | you get so many tickets per month |
| 22:42.59 | WIMPy | That's not new. |
| 22:43.07 | paule32 | how can i set a dialplan to call outside, i have read the documentation but can't find suitable informations, i don't have dahdi hardware |
| 22:43.08 | WIMPy | But I don't know if these sevices still exist. |
| 22:43.14 | monsterco | I gtg pick up gf now and head to a wedding reception - thanks for being patient with me guys - i will try the 4.0.4 split tomorrow |
| 22:43.27 | monsterco | WIMPy - this was a new startup |
| 22:43.44 | WIMPy | paule32: There is no inside and outside. There are just channels and peers. |
| 22:44.37 | paule32 | WIMPy, the html doc says SIP/YourVioCarrier |
| 22:44.50 | paule32 | but what can i understand under this |
| 22:45.51 | WIMPy | I take it you want to place calls to the PSTN? So do you have any connection to the PSTN? Either via some sort of hardware or an ITSP? |
| 22:46.59 | paule32 | i have use non public closed source application and can make calls from pc to outsode of it |
| 22:47.17 | paule32 | but how to set up this in asterisk, i don't know |
| 22:47.44 | WIMPy | What did that application connect to? Do yu have login data? |
| 22:47.46 | paule32 | i have a fritzbox dsl modem router that have a passthrough |
| 22:48.12 | monsterco | [TK]D-Fender - 2354-......sip something is not compatible with the phone - so 4.0.4 split seems to not be good for phone |
| 22:48.15 | WIMPy | So you want to connect Asterisk to the FB? |
| 22:48.17 | monsterco | so phone escaped updating |
| 22:48.40 | paule32 | login data is set on router, i can't show login data, all is crypted by this software |
| 22:48.46 | monsterco | i am out - later |
| 22:48.52 | *** part/#asterisk monsterco (~monsterco@64.231.101.21) |
| 22:48.54 | paule32 | WIMPy, yes |
| 22:49.01 | WIMPy | It's easy to decrypt. |
| 22:49.32 | WIMPy | Are you with a provider that doesn't give you the login data? |
| 22:50.06 | paule32 | i have register a tel. number, give it a own password |
| 22:50.22 | paule32 | do configure the router |
| 22:50.30 | WIMPy | Errr, what? |
| 22:50.42 | paule32 | the server is: fritz.box |
| 22:50.52 | paule32 | the user is 100 |
| 22:50.55 | paule32 | eg. |
| 22:51.10 | paule32 | and the password is: secret (e.g.) |
| 22:51.35 | WIMPy | Ok, back to that. Yes, you can make a peer in asterik with that data. |
| 22:52.09 | paule32 | cool, and how? |
| 22:52.25 | WIMPy | Look at your sip.conf. |
| 22:52.34 | paule32 | register => ....? |
| 22:52.39 | paule32 | that is done |
| 22:53.36 | WIMPy | That's the old way, but ok. |
| 22:53.57 | WIMPy | So you should be able to receive calls then. |
| 22:54.57 | paule32 | no, sorry |
| 22:55.46 | WIMPy | So are you registered? |
| 22:58.06 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 23:00.10 | paule32 | http://codepad.org/iUclWlQ6 |
| 23:00.13 | paule32 | yes |
| 23:01.49 | WIMPy | And where is the peer for your FB account? |
| 23:02.23 | *** join/#asterisk commandocoding (~commandoc@59.180.154.128) |
| 23:02.35 | commandocoding | Hello everyone |
| 23:02.47 | WIMPy | And BTW: Are 1234 and 4321 the real passwords? |
| 23:03.53 | paule32 | at the testing moment , yes |
| 23:04.09 | paule32 | ah i understand, when 100 is the user name |
| 23:04.14 | commandocoding | My windows install of asterisk detects the analogue modem, and also when I call my landline I do get some prompts etc. Now the question is, can I use asterisk to record the conversation while i pick the call from the analogue modem out phone set or a voip phone inside the network. |
| 23:04.31 | paule32 | then the peer 100 must exist (user name) ? |
| 23:04.33 | WIMPy | Should be something like 620 IIRC? |
| 23:04.45 | paule32 | right, indeed |
| 23:04.57 | WIMPy | yes |
| 23:05.09 | WIMPy | Or you allowguests. |
| 23:05.18 | paule32 | i don't know, that anything here came from germany |
| 23:05.25 | paule32 | nono :) |
| 23:05.45 | WIMPy | #asterisk-de |
| 23:06.39 | paule32 | hui so little amount of people there |
| 23:06.49 | paule32 | ok it is late here |
| 23:08.41 | paule32 | and the variable OUTBOUNDTRUNK=SIP/1.2.3.4 is ok? |
| 23:09.01 | WIMPy | Do you use it? |
| 23:09.01 | Penguin | no |
| 23:09.15 | Penguin | use the PEER NAME |
| 23:09.26 | paule32 | or must it be SIP/620 |
| 23:09.55 | WIMPy | Makes more sense, IF you want to use such a variable. |
| 23:10.27 | navaismo | slap the ~book |
| 23:10.49 | WIMPy | Or the samples? |
| 23:10.51 | WIMPy | Or both? |
| 23:10.56 | paule32 | i read the book and other sources |
| 23:10.57 | commandocoding | Guys can any one suggest a service where I can test asterisk, is there any hoisted asterisk server for testing available? |
| 23:11.01 | paule32 | [Sep 14 01:10:15] WARNING[32294]: channel.c:5755 ast_request: No channel type registered for '' |
| 23:11.01 | paule32 | [Sep 14 01:10:15] WARNING[32294]: app_dial.c:2218 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented) |
| 23:11.37 | WIMPy | What did you try to do? Dial nothing? |
| 23:12.26 | paule32 | 0 <village number> <house number> |
| 23:12.46 | WIMPy | In your dialplan. |
| 23:13.15 | paule32 | exten => _0X.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 23:13.15 | paule32 | <PROTECTED> |
| 23:14.08 | WIMPy | So OUTBOUNDTRUNK seems to be empty. |
| 23:14.20 | WIMPy | And why do you cut off the first digit? |
| 23:14.24 | navaismo | :@ |
| 23:14.58 | paule32 | the example shows it |
| 23:15.15 | WIMPy | Forget about that. |
| 23:15.15 | paule32 | here in germany we have 0 before each town number |
| 23:15.27 | WIMPy | Cut&paste configuration won't get you anywhere. |
| 23:15.27 | Penguin | If you are going to set the GLOBAL VARIABLE of OUTBOUNDTRUNK, make sure you set it in the globals section of the dial plan. |
| 23:16.10 | WIMPy | Yes, so why do you cut that 0? You need to send it to your telco. |
| 23:16.20 | WIMPy | (unless you use ISDN and set the correct TON) |
| 23:16.39 | Penguin | He probably has no idea what the EXTEN:1 thing means. |
| 23:16.47 | paule32 | oh my god, Penguin indeed, this makes sense |
| 23:16.51 | paule32 | im stupid |
| 23:16.54 | WIMPy | That's what I think. |
| 23:16.54 | paule32 | :/ |
| 23:17.08 | Penguin | And the fact that you have used the "same" keyword but still specified the extension... |
| 23:17.16 | paule32 | WIMPy, thank you :/ |
| 23:17.38 | Penguin | same => n,Congestion() |
| 23:17.42 | Penguin | not what you wrote. |
| 23:17.56 | WIMPy | And maybe not that sensible anyway. |
| 23:19.28 | Penguin | So I got an email earlier saying sipgate was discontinuing services and would be shutting down. |
| 23:19.38 | Penguin | Their web site still has services for sale, though. |
| 23:20.13 | WIMPy | Oh. I thought thay already had withdrawn from the US. |
| 23:20.33 | *** join/#asterisk commandocoding (~commandoc@106.204.179.30) |
| 23:23.49 | navaismo | commandocoding, no, as far i know. What do you want to test? |
| 23:30.38 | *** join/#asterisk commandocoding (~commandoc@106.204.179.30) |
| 23:31.53 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 23:31.54 | *** part/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 23:37.05 | paule32 | jippy |
| 23:37.17 | paule32 | can make outgoing calls |
| 23:37.36 | paule32 | SIP/620 was wrong |
| 23:37.45 | paule32 | the carrier was the router ip |
| 23:37.50 | paule32 | but |
| 23:37.53 | Penguin | Did you have an entry for [620] in the sip.conf? |
| 23:38.07 | paule32 | yes |
| 23:38.33 | Penguin | Then at least SIP/620 is valid. |
| 23:38.46 | Penguin | Even if the configuration for 620 was bad, it was a valid peer to call. |
| 23:39.32 | WIMPy | And the right one. |
| 23:39.44 | paule32 | so now, i have to gorward a port to asterisk? 5060? for speech? |
| 23:39.55 | paule32 | forward |
| 23:40.03 | Penguin | UDP 5060 for SIP. |
| 23:40.14 | WIMPy | You can't do that. |
| 23:40.24 | paule32 | the problem is, yes |
| 23:40.26 | paule32 | indeed |
| 23:40.30 | Penguin | UDP 10000-20000 for RTP (if that is the port range in rtp.conf) |
| 23:40.40 | WIMPy | You need to use another port. |
| 23:40.57 | Penguin | His modem has SIP enabled and captures the port? |
| 23:41.11 | WIMPy | yes |
| 23:41.11 | paule32 | yes |
| 23:41.24 | Penguin | Can you change that port and use the normal one for asterisk? |
| 23:42.03 | paule32 | you can, but then all phones are dead, when asterisk is not running |
| 23:42.12 | paule32 | or? |
| 23:42.40 | WIMPy | The question was if you can configure the router to use another port. |
| 23:43.00 | paule32 | i check it one moment please |
| 23:43.09 | WIMPy | I'm not sure, but I don't think you can without manually editing the config. |
| 23:48.27 | Penguin | My ITSP provides an alternate port for such cases. |
| 23:48.39 | Penguin | Actually, more than one alternate port. |
| 23:48.48 | WIMPy | Wrong end? |
| 23:49.37 | Penguin | Oh maybe. I assume if I use the alternate port they will send calls to me on the alternate port as well. |
| 23:50.05 | WIMPy | Hopefully not. |
| 23:50.22 | Penguin | If my modem was stopping 5060 from getting to my asterisk, I would want them to send calls to me on another port. |
| 23:50.32 | paule32 | it can be done with "exposed host" |
| 23:50.43 | WIMPy | Yes, but you tell them when registering. |
| 23:50.44 | paule32 | it seems it stands for dmz |
| 23:51.04 | Penguin | Oh, right. I wasn't thinking registration. |
| 23:51.05 | WIMPy | Your router is listening on 5060. You can't forward that port. |
| 23:51.21 | Penguin | Do not use DMZ when configuring asterisk. |
| 23:51.57 | paule32 | no other choice Penguin |
| 23:52.07 | Penguin | Of course there is. |
| 23:52.29 | Penguin | DMZ is not an alternative to proper port forwarding. |
| 23:52.52 | Penguin | And if your router/modem is taking away 5060, even a real DMZ wouldn't help. |
| 23:57.10 | *** join/#asterisk spengler1 (~spengler@pool-98-117-213-86.bltmmd.fios.verizon.net) |
| 23:57.32 | spengler1 | whats the best way to handle calls coming into a queue |
| 23:57.50 | spengler1 | my dial plan looks like this |
| 23:57.54 | spengler1 | exten => 7001,1,Answer |
| 23:57.54 | spengler1 | exten => 7001,2,Ringing |
| 23:57.54 | spengler1 | exten => 7001,3,Wait(2) |
| 23:57.56 | spengler1 | exten => 7001,4,Queue(support) |
| 23:57.58 | spengler1 | exten => 7001,5,Dial(SIP/2002,20) |
| 23:58.00 | spengler1 | exten => 7001,6,Voicemail(2002@default&2001@default) |
| 23:58.01 | WIMPy | ~pb |
| 23:58.02 | 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. |
| 23:58.02 | spengler1 | exten => 7001,7,Wait |
| 23:58.04 | spengler1 | exten => 7001,8,Hangup |
| 23:58.11 | spengler1 | sorry |
| 23:58.47 | [TK]D-Fender | spengler1: there is no such thing as "best" |
| 23:58.49 | spengler1 | so the problem with this is that if one of my agents is paused the logic goes to the next step to dial sip/2002 because |
| 23:58.54 | [TK]D-Fender | spengler1: there is on "what do YOU want". |
| 23:59.04 | WIMPy | What's that Answer;Ringing;Wait crap? |
| 23:59.35 | spengler1 | i probably pulled that off an example somewhere |
