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| 00:28.35 | Spengler1 | Hellooooo everybody |
| 00:29.25 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 00:33.39 | Spengler1 | anyone in here use broadvoice |
| 00:34.34 | navaismo | not me |
| 00:34.53 | Spengler1 | any recommendations for a voip service? |
| 00:38.27 | *** join/#asterisk fetalbird (~fetalbird@184.82.177.75) |
| 00:38.34 | navaismo | ~usitsp-list |
| 00:38.43 | navaismo | ~itspus-list |
| 00:38.49 | navaismo | ~itsp |
| 00:38.49 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 00:38.55 | navaismo | ok was |
| 00:39.05 | navaismo | ~itsplist-us |
| 00:39.05 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 00:39.13 | navaismo | Spengler1, ^^ |
| 00:40.24 | gundy | teliax? huh. I've had problems with them. |
| 00:40.33 | gundy | I'm much rather go with flowroute. |
| 00:40.51 | gundy | Anyway. Not trying to stir things up. |
| 00:45.24 | Spengler1 | what are your thoughts on deploying these services to businesses? |
| 00:46.06 | navaismo | You need a good infrastructure a good QoS network |
| 00:46.40 | Spengler1 | i'm segmenting the voip onto a seperate vlan ; is that sufficient for qos or should i do more? |
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| 01:04.47 | jmetro | depends |
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| 01:12.38 | jmetro | oh lawd. |
| 01:12.47 | jmetro | has anyone here ever put SDP on asterisk |
| 01:13.37 | jmetro | my valcom unit just provoked the error "NOTICE[1369][C-00000428]: chan_sip.c:10453 process_sdp: No compatible codecs, not accepting this offer! |
| 01:13.37 | jmetro | " |
| 01:16.43 | navaismo | well that said a lot, you need to use same codecs to made the call |
| 01:17.07 | jmetro | hm. That makes me realize that i have no idea what codec this thing is using. |
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| 01:18.31 | jmetro | aha...g711. |
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| 03:59.11 | Doozer_ | hi all, i've managed (i think) to register 3 extensions, but don't seem to be able to call from one extension to another (yet I can make outbound calls). Looking for some help please :) |
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| 04:32.56 | ChannelZ | Can you be more vague? |
| 04:33.09 | ChannelZ | Did you write any extensions in extensions.conf to make them call each other? |
| 04:34.22 | Doozer_ | I can be, but i doubt it'll help |
| 04:34.43 | Doozer_ | i have added 3 extensions, 2 real phones, one soft |
| 04:34.49 | Doozer_ | the soft is able to call etc fine |
| 04:35.01 | Doozer_ | the real ones show up as status UNKNOWN |
| 04:35.13 | Doozer_ | which i suspect means they aren't registering properly |
| 04:37.50 | Doozer_ | ok to paste a sip show peer xxx to the chan? |
| 04:49.48 | fetalbird | put the sip and extensions in a paste, ill check it out |
| 04:49.59 | fetalbird | its prob just wrong contexts |
| 04:50.44 | Doozer_ | great thanks, be a couple of mins |
| 04:56.12 | Doozer_ | http://pastebin.com/81rYacit |
| 04:56.18 | Doozer_ | let me know if i missed something |
| 05:01.12 | fetalbird | Those configs are for freepbx, not for a vanilla install |
| 05:01.26 | Doozer_ | dialing *61 gives me "The number you have called is not in service" |
| 05:01.37 | Doozer_ | correct |
| 05:01.41 | ChannelZ | Thank god I wandered off to go poop |
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| 05:03.47 | fetalbird | I cant solve your issue, someone else might or try #freepbx |
| 05:04.33 | Doozer_ | ok, will do, thanks for taking a look |
| 05:04.42 | fetalbird | no prob |
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| 05:12.37 | fabiobik | hello guys |
| 05:12.55 | fabiobik | ive installed on vm the iso at asterisk.org |
| 05:15.08 | ChannelZ | Ok |
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| 08:32.54 | slicknick5181 | asterisk 1.8 my console commands have disappeared from my cli help list |
| 08:33.07 | slicknick5181 | and none of the commands work |
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| 09:47.48 | *** join/#asterisk rasheed (~rasheed@bba164979.alshamil.net.ae) |
| 09:48.06 | rasheed | hello |
| 09:48.21 | rasheed | i need a help regarding queue |
| 09:48.26 | rasheed | can some one help me |
| 09:49.48 | rasheed | any body here |
| 09:50.36 | *** part/#asterisk rasheed (~rasheed@bba164979.alshamil.net.ae) |
| 09:51.16 | *** join/#asterisk rasheed (~rasheed@bba164979.alshamil.net.ae) |
| 09:51.24 | rasheed | hello |
| 09:51.29 | rasheed | some body there? |
| 09:53.17 | rasheed | i would like to discuss a strange behaviour of queue .. |
| 09:54.23 | kaldemar | ~ask |
| 09:54.23 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:55.57 | rasheed | we have deployed asterisk 1.8 in a site. and i configured queue with ringinuse=no |
| 09:56.29 | rasheed | but i noticed that , randomly some times 2 calls are still routing to same extension |
| 09:56.34 | rasheed | extensions are all SIP |
| 09:57.08 | rasheed | i was not able to regenerate this case, when testing with only 2 trunks |
| 09:57.32 | kaldemar | by extensions do you mean queue members? |
| 09:57.33 | rasheed | in the deployed site, there are 13 trunks and 32 extensions |
| 09:57.39 | rasheed | yes queue members |
| 09:57.41 | kaldemar | extensions are something in your dialplan. |
| 09:57.49 | rasheed | sorry i mean queue members |
| 09:58.09 | rasheed | i add dynamic queue members using addqueuemember |
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| 09:59.19 | kaldemar | do you have callcounter=yes configured in sip.conf for the peers that are members in your queue? |
| 09:59.56 | rasheed | in fact.. let me explain my configuration.. |
| 10:01.15 | kaldemar | you can also show the configurations in pastebin instead of trying to explain what you have configured. |
| 10:01.51 | rasheed | i will try to do that.. but i am new to IRC.. let me check the options.. |
| 10:03.32 | kaldemar | ~pb |
| 10:03.32 | 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. |
| 10:03.43 | kaldemar | but first, answer the question about callcounter. |
| 10:03.50 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html |
| 10:04.03 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html#ACD_id288726 |
| 10:04.05 | rasheed | yes callcounter =yes for all the SIP extensions.. defined |
| 10:06.29 | rasheed | the same sip interfaces are defined for addqueuemember |
| 10:12.58 | rasheed | i pasted part of my configuration in pastebin |
| 10:13.56 | kaldemar | feel free to paste the link to your pastebin here. |
| 10:14.01 | rasheed | ok .. |
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| 10:14.48 | rasheed | http://pastebin.com/YnHTpKhL |
| 10:15.08 | rasheed | this is for queue member 1040 |
| 10:17.38 | rasheed | http://pastebin.com/zvN1i5UH |
| 10:17.47 | rasheed | this is the login with addqueuemember |
| 10:19.40 | rasheed | the calls from trunk will be directly passed to the queue .. and if there are free agents who logged in using addqueuemember, call will be routed to him |
| 10:20.50 | kaldemar | i don't see anything passing directly to any queue. |
| 10:21.10 | rasheed | i will paste that section now.. |
| 10:21.14 | rasheed | please wait |
| 10:22.54 | kaldemar | anyway, the callcounter won't help you much since you don't use the sip interfaces as members, but local channels. |
| 10:23.36 | rasheed | oh ok.. |
| 10:23.51 | rasheed | is there any other way |
| 10:24.04 | kaldemar | i just mentioned two ways. pick one. |
| 10:24.19 | rasheed | if i use sip interfaces as members,, i think it is not possible to have free seatring for agents right? |
| 10:24.47 | rasheed | i am sorry if i am asking some thing wrong.. but i am very new .. and i really appreciate your great feedbacks |
| 10:25.12 | kaldemar | clarify "to have free seatring for agents" |
| 10:26.20 | rasheed | i mean , the agents work on shift basis.. they need to login to the phone to get the calls routed to them.. |
| 10:26.31 | rasheed | if i use fixed sip interfaces as members.. i dont know how it is possible |
| 10:27.38 | kaldemar | not fixed. use the queue member applications just like you do now, but don't use local channels as the interface. use the SIP/... directly. |
| 10:29.56 | rasheed | do you mean.. in addqueuemember command, the interface should be SIP/104 like this ? |
| 10:30.11 | kaldemar | something like that, yes. |
| 10:30.29 | rasheed | in fact it is done like that , in the dial plan |
| 10:30.34 | rasheed | if you have noticed |
| 10:31.49 | kaldemar | i have not. the only AddQueueMember line in your configs is exten => 701,n,AddQueueMember(7000,local/${AUTH_MAILBOX}@agents/n... |
| 10:32.08 | rasheed | if you check the dial plan for agent login with addqueuemember.. the last parameter was the interface.. where i put SIP/CALLERID(number) which will be the person who calls the dialplan |
| 10:32.27 | rasheed | i will paste the complete line |
| 10:32.33 | rasheed | please wait.. |
| 10:32.35 | kaldemar | don't bother, i see the line. |
| 10:32.39 | rasheed | ok. |
| 10:32.52 | kaldemar | the syntax is AddQueueMember(queuename[,interface[,penalty[,options[,membername[,stateinterface]]]]]) |
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| 10:33.44 | kaldemar | okay, you do have the stateinterface, that one i missed. |
| 10:33.59 | rasheed | yes.. |
| 10:35.38 | rasheed | i am using leastrecent strategy |
| 10:37.41 | kaldemar | what you have yet to show is how the calls get to the queue in the first place and an actual call. |
| 10:37.52 | rasheed | ok.. i am taking that |
| 10:37.56 | rasheed | please wait.. |
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| 10:45.31 | rasheed | http://pastebin.com/fQUH69ZR |
| 10:45.35 | rasheed | please check this.. |
| 10:46.03 | rasheed | i will paste the queues.conf also |
| 10:48.32 | rasheed | queues.conf is also pasted in http://pastebin.com/UMPHRKvt |
| 10:53.47 | kaldemar | and the call... |
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| 11:10.28 | rasheed | do you need the call log? |
| 11:22.07 | rasheed | hello |
| 11:22.19 | rasheed | i will send the sample call log of a file |
| 11:24.01 | rasheed | http://pastebin.com/4t58EHBb |
| 11:24.13 | rasheed | please have a look at this call log.. where 2 calls routed to same agent |
| 11:25.40 | rasheed | i will be back in 5 minutes |
| 11:29.37 | kaldemar | "SIP_STATE=NOT_INUSE" and "SIP/101-00000042 is busy" don't quite match. |
| 11:33.31 | kaldemar | your sip configs may be to blame. |
| 11:39.01 | rasheed | oh ok |
| 11:39.34 | rasheed | do you need the SIP configs, i can paste it |
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| 12:03.28 | rasheed | http://pastebin.com/9TwkfSfL |
| 12:03.37 | rasheed | i pasted the sip_additional.conf |
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| 12:13.38 | rasheed | hello kaldemar |
| 12:14.00 | rasheed | did u get the pastebin of the sip conf |
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| 12:36.50 | kresp0 | rasheed, your paste has expired |
| 12:42.15 | rasheed | i will paste again |
| 12:43.51 | rasheed | http://pastebin.com/javQh7Kt |
| 12:43.56 | rasheed | i have a question.. |
| 12:44.08 | rasheed | in the previous log of call logs which i posted.. |
| 12:44.45 | rasheed | you can see that queue is sending 2 calls to local interface 2030 from dahdi channel 6 and 12.. |
| 12:45.07 | rasheed | in the normal scenario queue application should not do that , i believe .. is that right |
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| 12:46.03 | rasheed | provided ringinuse=no.. |
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| 12:46.56 | rasheed | i am not sure , how the queue internally track calls.. do you think if it is tracking the call based on SIP interface status?.. |
| 13:09.47 | kresp0 | rasheed, sorry, I'm new to queues. But I think that line 6 on http://pastebin.com/fQUH69ZR have at least several syntax errors. |
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| 14:06.24 | rasheed | <rasheed> hello |
| 14:06.25 | rasheed | <rasheed> sorry my internet connection was lost in between |
| 14:06.25 | rasheed | <rasheed> can you please type the last message again , regarding the syntax errors |
| 14:06.25 | rasheed | <rasheed> i just lost it |
| 14:18.19 | leifmadsen | rasheed: http://ofps.oreilly.com/titles/9781449332426/asterisk-ACD.html |
| 14:18.45 | leifmadsen | if you're having issues with sip channels not having the appropriate device state, there are thigns you need to enable |
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| 14:27.04 | rasheed | ok thank you .. |
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| 14:44.29 | KNERD | What are the list of CFLAGS, etc supported by the config script? |
| 14:45.35 | KNERD | I am dealing with this crap https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 14:45.36 | LieutPants | [ASTERISK-20128] [Status: Open] [Assigned: kmoore] Virtualized asterisk.org 1.8.14.0 no longer runs in a KVM virtualized environment. Compiles without error, but fails with Illegal instruction on launch Regression since 1.8.13.0 Last good 1.8.12.2 - https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 14:47.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:55.12 | KNERD | What about using an Intel compiler? |
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| 15:46.59 | igcewieling | Love this "[Mar 23 11:46:06] NOTICE[12358]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.119.18" .18 is an ASTERISK server. |
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| 16:04.23 | kchehab | hi i use to add |
| 16:04.24 | kchehab | exten => conn,n,Dial(SIP/Comp_220/${number}:3,30) |
| 16:04.24 | kchehab | exten => conn,n,Dial(SIP/Comp_221/${number}:3,30) |
| 16:05.01 | kchehab | but when call fails on Comp_220/ its not jumping to Comp_221 |
| 16:05.07 | kchehab | any advice |
| 16:05.50 | [TK]D-Fender | kchehab: look at the call. |
| 16:09.02 | kchehab | sorry ${number:} |
| 16:09.05 | kchehab | sorry ${number:3} |
| 16:09.51 | [TK]D-Fender | kchehab: I was wondering if that was a variable mistake.... |
| 16:09.53 | kchehab | [TK]D-Fender is there a way to make a load balance between these to gateways |
| 16:10.06 | kchehab | two* or three |
| 16:10.52 | [TK]D-Fender | kchehab: define "load balance" |
| 16:12.30 | kchehab | [TK]D-Fender i have three trunks and i want to let the calls go to the three trunk with percentage not to make an order 1st 2nd 3rd ,sicne the 3rd will have a low ASR |
| 16:13.30 | igcewieling | kchehab: you will have to manually code that in the dialplan |
| 16:13.42 | [TK]D-Fender | ^ |
| 16:15.09 | igcewieling | of all your peers are Comp_22x where x = 0-2 exten => conn,n,Dial(SIP/Comp_22${RAND(0,2)}/${number:3},30) but that does NOT do failover |
| 16:15.15 | igcewieling | s/of/if |
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| 17:17.42 | igcewieling | Heh, no calls, no calls, no calls, BAM! 300 calls from one customer. Expected, but still interesting. |
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| 17:53.01 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:06.16 | *** join/#asterisk rasheed (rasheed@bba402370.alshamil.net.ae) |
| 18:22.53 | *** join/#asterisk slidesinger-lt (~jtatum@c-69-141-208-250.hsd1.nj.comcast.net) |
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| 18:52.06 | rasheed | hello |
| 18:52.58 | rasheed | if 2 calls hit a queue at the same time , is there a chance to route the call to same sip extension even when ringinuse =no |
| 18:53.27 | [TK]D-Fender | virtually impossible |
| 18:54.13 | rasheed | i got this behaviour with a system of 13 FXO lines and 32 extensions |
| 18:54.43 | rasheed | the logs are posted in http://pastebin.com/4t58EHBb for this behaviour |
| 18:54.44 | [TK]D-Fender | Analog is already very slow to get answered. the odds of those coinciding is incrementally lower |
| 18:56.00 | rasheed | as given in the logs , it is detecting the SIP_STATE as NOT_IN_USE even when it route another call to the same extension.. |
| 18:56.18 | igcewieling | do you have a call-limit set? |
| 18:56.19 | rasheed | i am using local interfaces as queue members.. |
| 18:56.36 | rasheed | in the sip extension , callcounter=yes |
| 18:56.42 | [TK]D-Fender | rasheed: what SIP state? we have no idea what that dialplan is doing |
| 18:57.10 | [TK]D-Fender | rasheed: -- Executing [2030@agents:2] Set("Local/2030@agents-0000002f;2", "SIP_STATE=NOT_INUSE") in new stack <- and this is DIALPLAN level. The call is already being DISTRIBUTED. It's too late |
| 18:57.37 | [TK]D-Fender | rasheed: If you didn't want to call the agent.. you need to tell the member what state device to use, not try to check it yourself in the dialplan. |
| 18:58.07 | rasheed | ok |
| 18:58.15 | [TK]D-Fender | rasheed: -- Executing [2030@agents:4] Playback("Local/2030@agents-0000002f;2", "agent_2030") in new stack ,- also, playing back to the client answers the queue call preventing further distribution or agent dials. This is shooting yourself in the foot |
| 18:59.08 | rasheed | i mentioned state interface in the addqueuemember |
| 18:59.24 | rasheed | please refer the post http://pastebin.com/zvN1i5UH |
| 18:59.53 | rasheed | so i expected that the queue will check the status of the sip interface before it is distributed |
| 19:00.19 | [TK]D-Fender | rasheed: "core show application addqueuemember" <- PB |
| 19:01.12 | rasheed | sorry i didnt get you.. |
| 19:01.41 | [TK]D-Fender | rasheed: PASTEBIN the instructions if give you |
| 19:01.54 | [TK]D-Fender | rasheed: Also we need to see your queue dump to see what you've actually added |
| 19:03.23 | rasheed | queue dump.. where can i take this.. |
| 19:03.48 | [TK]D-Fender | rasheed: so you never actually looked at your queue in CLI? |
| 19:04.03 | rasheed | i ddid that .. |
| 19:04.13 | rasheed | queue show.. i was checking |
| 19:04.29 | rasheed | do you want to PB queue show |
| 19:04.56 | rasheed | currently few other agents are logged in |
| 19:05.17 | [TK]D-Fender | show the whole queue. and the instructions page. |
| 19:05.26 | rasheed | ok |
| 19:05.29 | [TK]D-Fender | Also the dialplan for your agent dial-outs |
| 19:06.37 | rasheed | ok.. |
| 19:07.14 | rasheed | dial plan for agent dialouts is at http://pastebin.com/YnHTpKhL |
| 19:10.39 | rasheed | http://pastebin.com/fBNdjEFa |
| 19:10.50 | rasheed | i dump the queue and other details here.. |
| 19:10.58 | [TK]D-Fender | that dialplan for the agents appears to be for one agent, and not the one we actually dialed |
| 19:11.34 | [TK]D-Fender | exten => 1040-NOANSWER,n(hngup),Queue(7000,tr,,,10) <- and looking from 1 queue to another in a single dial? this looks like a very flawed design |
| 19:12.05 | rasheed | its the same for the other agent.. only difference is that instead of 1040, 2030 is the agent in the logs |
| 19:12.31 | [TK]D-Fender | I'd still like to see bits that at least match |
| 19:12.39 | [TK]D-Fender | And your queue shows all paused. |
| 19:12.44 | [TK]D-Fender | what happens with calls now? |
| 19:13.11 | rasheed | in fact.. the logs which i posted before was taken duringa test session.. of the call centre.. |
| 19:13.31 | rasheed | since we faced this issue.. we revert the lines back in to a previous pbx |
| 19:14.34 | [TK]D-Fender | You need to show us matching and relevant confgs & debug if we're to get anywhere |
| 19:15.03 | rasheed | ok.. |
| 19:15.38 | [TK]D-Fender | rasheed: exten => 1040,n(norm),Playback(agent_1040) - and this causes that first queue to always consider the agent dial as answered... |
| 19:15.39 | [TK]D-Fender | this is bad... |
| 19:16.00 | rasheed | oh ok.. |
| 19:16.06 | [TK]D-Fender | it will never distribute to another agent. |
| 19:16.07 | rasheed | we shall disable this.. |
| 19:16.44 | [TK]D-Fender | Are any of your agents processed differently than any other? |
| 19:17.11 | rasheed | sorry, i didnt get you .. |
| 19:17.38 | [TK]D-Fender | [agents] exten => 1040,1,Set(AGENT_SIP=${DB(agent_sip/1040)}) |
| 19:17.42 | [TK]D-Fender | we see you dialing 2030. |
| 19:17.51 | [TK]D-Fender | is 2030's call-flow any different from this one? |
| 19:18.02 | rasheed | no its the same |
| 19:18.10 | [TK]D-Fender | except for substituting 1040 for 2030? |
| 19:18.17 | [TK]D-Fender | then you should be using a pattern for this |
| 19:18.17 | rasheed | yes .. |
| 19:18.30 | [TK]D-Fender | so you don't risk making a tiny bug fixing one, but not all of them |
| 19:18.55 | rasheed | i am following same pattern for this |
| 19:19.00 | [TK]D-Fender | And using labels in that same pattern and not Goto another extension on status, etc |
| 19:19.07 | rasheed | for all agents |
| 19:19.15 | [TK]D-Fender | No, I'm saying your EXTENSION should be a PATTERN |
| 19:19.18 | [TK]D-Fender | not a fixed value |
| 19:19.28 | rasheed | oh ok.. |
| 19:19.45 | [TK]D-Fender | you must have a massive amount of dialplan code duplication for this by now |
| 19:19.51 | rasheed | but these are the agent id which the users log in |
| 19:20.00 | rasheed | through thier phones |
| 19:20.07 | *** join/#asterisk skrusty (~support@168.63.14.171) |
| 19:20.15 | rasheed | so the numbers will be different for each user |
| 19:20.21 | [TK]D-Fender | I'd suggest you clean all of that up. Clear out your memebers. get them back in and come back with a new matching scenario if there is anything in need of debugging at that point |
| 19:20.50 | [TK]D-Fender | yes, well the PROCESSING is identical. |
| 19:20.59 | rasheed | ok.. |
| 19:21.05 | [TK]D-Fender | [agents] exten => _XXXX,1,Set(AGENT_SIP=${DB(agent_sip/${EXTEN})}) |
| 19:21.07 | [TK]D-Fender | ^^^ |
| 19:21.17 | [TK]D-Fender | usage of patterns is Asterisk 101 |
| 19:21.20 | *** join/#asterisk Beta2K (~Beta2K@d24-36-163-88.home1.cgocable.net) |
| 19:21.33 | rasheed | ok.. |
| 19:21.35 | [TK]D-Fender | why have 30 lines repeated for 10 agents? the exten you dial IS the variable part |
| 19:21.44 | Beta2K | Hello all :) |
| 19:21.59 | rasheed | ok.. |
| 19:22.15 | rasheed | when adding queue member, i am using local interface |
| 19:22.31 | [TK]D-Fender | blocks Beta2K with a 4 inch wall of water |
| 19:22.43 | [TK]D-Fender | rasheed: Still fine so far... |
| 19:22.49 | rasheed | like Local/2030@agents/n |
| 19:23.21 | rasheed | so will that be granted , if i use patterns.. |
| 19:23.28 | [TK]D-Fender | rashYes and we can see you made THAT function variable as well.. this time using the CALLERID |
| 19:23.49 | rasheed | ok |
| 19:23.52 | [TK]D-Fender | All the values would seem to HAVE to work based on the circumstances of their creation. |
| 19:24.21 | rasheed | ok.. |
| 19:24.24 | [TK]D-Fender | You added 2030 as the exten in your local channel... no reason you can't take that at face value in the pattern match |
| 19:24.50 | Beta2K | Gee thanks Fender :) |
| 19:24.50 | rasheed | ok got it |
| 19:25.12 | rasheed | i will try to clean up |
| 19:27.09 | rasheed | do u think that ,Playback("Local/2030@agents-0000002f;2", "agent_2030") will make an issue identifying that 2030 is answered and queue will make again make another call to same 2030? |
| 19:28.04 | fetalbird | it will match a _XXXX in that context |
| 19:28.17 | rasheed | ok fine |
| 19:29.36 | [TK]D-Fender | rasheed: it DOES answer it |
| 19:29.44 | [TK]D-Fender | Playback ASNWERS calls |
| 19:29.49 | rasheed | ok.. |
| 19:29.57 | [TK]D-Fender | and you are in your local channel. |
| 19:30.48 | rasheed | ok..so there is a chance of queue considering the local channel as free , after the answer and route again another call? |
| 19:33.09 | [TK]D-Fender | Shouldn't.... |
| 19:33.19 | [TK]D-Fender | Actually.. |
| 19:33.25 | [TK]D-Fender | it is checking the DEVICE.... |
| 19:33.29 | rasheed | ok |
| 19:33.34 | [TK]D-Fender | which is a SIP phone that hasn't even been DIALED yet |
| 19:33.36 | [TK]D-Fender | so YES |
| 19:33.44 | [TK]D-Fender | while that pacyback is happening... you COULD get another call |
| 19:33.48 | [TK]D-Fender | All part of the same failure |
| 19:33.53 | rasheed | ok... |
| 19:34.05 | [TK]D-Fender | And Requeuing the call.... bad idea |
| 19:35.20 | rasheed | ok.. we were trying to requeue becuase, some times agents put thier handset down to escape from calls.. in this case , the call will keep on hitting the same agent until he put the handset back.. |
| 19:35.34 | rasheed | so we made the agent to pause.. and then requeue in this situation |
| 19:36.29 | Beta2K | Anyone around tried to get a Polycom 8002 to register? I'm seeing these incomming SIP register packets, but asterisk isn't sending anything back? http://pastebin.com/FHAbBHUX |
| 19:36.34 | [TK]D-Fender | Putting a handset down after answering.... doesn't end the call. |
| 19:37.17 | rasheed | it will not end the call.. but will keep trying same agent again and again and finally he get rid of that.. |
| 19:37.32 | rasheed | it is an emergency call centre.. so they dont want to loose calls.. |
| 19:39.31 | rasheed | if this is dangerous to do like that , i will try to avoid that |
| 19:39.38 | [TK]D-Fender | If yo have an agent who isn't answering... and all you are going to do is hammaer thta one guy ... then your CALLER isn't getting answered for their emergency and you aren't helping the caller You are hurting their ability to GET an answer from someone who isn't slacking. |
| 19:39.56 | [TK]D-Fender | Simply LOG it and pause them and send a notice to a manager |
| 19:40.07 | igcewieling | rasheed: the solution to that problem is when an agent makes the call go away, have the call be transfered to the person's boss. |
| 19:40.08 | rasheed | ok... |
| 19:40.21 | rasheed | ok. |
| 19:40.21 | [TK]D-Fender | But you are preventing these apparent emegerncy callers from getting actual service and making a mess of your PBX while doing that |
| 19:40.32 | rasheed | ok i got it.. |
| 19:40.53 | rasheed | so i will try to take away.. the playback.. , requeueing from the dial plan |
| 19:41.30 | [TK]D-Fender | I don't see a reason to requeue |
| 19:41.41 | rasheed | ok.. will take it away |
| 19:43.16 | rasheed | is it ok to route it to another queue instead? |
| 19:43.44 | [TK]D-Fender | Why do you need another queue? |
| 19:44.10 | [TK]D-Fender | rasheedis it ok to route it to another queue instead? <- this IS sending it to another queue which I just asked WHY you felt the need for... |
| 19:45.07 | rasheed | hmm.. ok , i dont think it is required.. |
| 19:45.19 | rasheed | will try to avoid any requeuing to make it simple |
| 19:46.14 | *** join/#asterisk bdunn (~bdunn@cpe-173-175-208-179.tx.res.rr.com) |
| 19:46.35 | bdunn | Hi - can anyone tell me how to contact the 24/7 support from Digium? |
| 19:46.51 | [TK]D-Fender | www.digium.com <- |
| 19:46.58 | bdunn | I've searched there. |
| 19:47.14 | igcewieling | I was not aware Digium provided 24/7 support. |
| 19:47.17 | rasheed | thanks a lot fender.. i will get back to you with the results... |
| 19:47.22 | [TK]D-Fender | bdunn: http://www.digium.com/en/company/contact <-- |
| 19:47.28 | bdunn | http://www.digium.com/en/products/asterisk/support |
| 19:48.09 | [TK]D-Fender | bdunn: "A: Support is available, depending on your agreement, twenty-four (24) hours a day, around the clock." <-- and your agreement offers this? |
| 19:48.13 | bdunn | They have commercial support, which I have, but I cannot find the telephone number that will take me to a real live person. |
| 19:48.14 | igcewieling | heh, interesting. Though it does say "per your afreement" |
| 19:48.44 | igcewieling | For some reason I thought they only offered support contracts for their commercial Asterisks. |
| 19:48.46 | bdunn | Oh wait… the phone tree is different this weekend. |
| 19:49.19 | bdunn | Damn - same result… "Our office is currently closed" |
| 19:49.44 | bdunn | I have several 24/7 support contacts but I have never needed it before. |
| 19:50.59 | [TK]D-Fender | You should have a number from them with it... |
| 19:53.30 | bdunn | Worked my way through the tree to support and it asked for a phone number to the account. I put in about five and none are recognized. Oh well. So much for 24/7 support. |
| 19:54.11 | bdunn | I guess I need to setup a new Asterisk box quickly and try to find a solution that way. Any suggestions for doing that quickly? Trixbox or FreePBX or what else is there? |
| 19:54.57 | [TK]D-Fender | depends what you need & want |
| 19:55.29 | bdunn | Basically I need to test a few polycom phones against a system. I have a Switchvox from Digium which is having issues and I need to rule out the phones. |
| 19:55.51 | bdunn | I'm about 50 hours into this counting last weekend and this weekend. Need to start isolating the issue. |
| 19:56.39 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 19:56.51 | bdunn | AsteriskNow? |
| 19:57.19 | Beta2K | Don't even bother looking at Trix |
| 19:57.26 | Beta2K | It hasn't been updated in over two years |
| 19:57.30 | [TK]D-Fender | 3 |
| 19:57.33 | [TK]D-Fender | and never will be |
| 19:57.34 | [TK]D-Fender | it is dead |
| 19:57.52 | [TK]D-Fender | FreePBX ISO or AsteriskNOW are the best ISO options so far |
| 19:57.55 | bdunn | Good to know. FreePBX, AsteriskNow? Anything else? |
| 19:58.04 | Beta2K | I'm in the process of trying to decide, Elastix or PIAF |
| 19:58.38 | rasheed | i think elastix will be the best |
| 19:58.39 | [TK]D-Fender | Beta2K: Nethier.... Elastix is embedded FreePBX and broken crap to support |
| 19:58.46 | [TK]D-Fender | PIAF bundles too much extra junk. |
| 19:59.23 | tm1000 | also there's elastic v2 which does what [TK]D-Fender said and v3 which is a rewrite not using freepbx, but has limited support for well…anything |
| 19:59.23 | Beta2K | That's what I've been noticing :) I don't even login to Elastix front end anymore, straight to freepbx |
| 19:59.41 | [TK]D-Fender | If you're spending your time working around it... you are making a mistake |
| 20:00.28 | bdunn | Any opinions on FreePBX vs. AsteriskNow? |
| 20:00.33 | rasheed | i am not using elastix call centre and other modules |
| 20:00.40 | rasheed | i just use the free pbx in it |
| 20:01.14 | [TK]D-Fender | bdunn: AsteriskNOW just got a new release. Not sure how is measures up. But the FreePBX ISO has been solid for along time and the GUI is directly supported |
| 20:02.11 | bdunn | Looks like there is a lot more web based information for FreePBX. Think I'll go with that. I'm concerned that AsteriskNow will interfere with Digium's Switchvox business which means it may be crippled somewhat. |
| 20:02.16 | tm1000 | rasheed: If you have no need for call centre or anything BUT freepbx then elastic is NOT your best choice. |
| 20:02.27 | Beta2K | Ugh, who would have thought a 3com phone would be easier to setup then a polycom... |
| 20:02.34 | rasheed | oh ok.. |
| 20:02.48 | [TK]D-Fender | bdunn: What may be crippled? |
| 20:03.10 | bdunn | AsteriskNow. |
| 20:03.26 | [TK]D-Fender | it isn't |
| 20:03.30 | rasheed | yes this is one concern why i didnt go for asterisk now |
| 20:03.33 | [TK]D-Fender | It is exactly what it looks like |
| 20:04.09 | rasheed | ok |
| 20:04.47 | bdunn | While I'm here… I have Polycom IP phones that are retaining settings from a different SwitchVox (asterisk appliance) system. No matter what I do the settings keep coming back and I have no idea how. Anyone have any ideas? |
| 20:05.29 | [TK]D-Fender | they are pointed to a provisioning server |
| 20:05.30 | Beta2K | Any idea why asterisk isn't responding to a sip register request? I see it coming in on the cli, but it doesn't send anything back to the phone? |
| 20:05.30 | [TK]D-Fender | change it |
| 20:05.46 | Beta2K | bdunn, what model phone? |
| 20:05.50 | bdunn | 650 |
| 20:06.43 | bdunn | I get on the web server for the phone and make the changes to SIP or Lines and it will reboot with the same settings. I have tried using the phone menu to reset local config, reset settings, erase firmware. |
| 20:07.05 | bdunn | The old system isn't on this network - only the new one, and it has none of this config on it. It's a new system. |
| 20:07.55 | bdunn | Tried 4,6,8,* |
| 20:07.55 | [TK]D-Fender | And did you disable provisioning so it can't detect it or try to contact it in the BootROM? |
| 20:08.10 | Beta2K | if it can't pickup files from the provisioning server it will used it's stored ones |
| 20:08.15 | bdunn | I guess I don't know how to do that. |
| 20:08.35 | Beta2K | even if you set it up different in the menu |
| 20:08.48 | bdunn | Beta2K: That may be what is happening. Here's the weird thing. NEW 650 phones are just fine. Those that were brought from another site will not lose their old settings. |
| 20:09.23 | Beta2K | Provision them, I've got a couple 430's I have a similar issue with |
| 20:09.39 | Beta2K | You could also try updating their FW to force them to erase their flash |
| 20:10.02 | [TK]D-Fender | Restart phone. Enter BootROM BEFORE it boots. kill of provisioning. |
| 20:10.11 | bdunn | That's what I was going to try next with another phone system like FreePBX as Switchvox doesn't give you a lot of options there. |
| 20:10.12 | [TK]D-Fender | point to NEW place for it to look |
| 20:10.34 | [TK]D-Fender | set up FTp and drop a stock firmware pack there |
| 20:10.47 | [TK]D-Fender | You don't even need to isntall Asterisk to just fix your phones |
| 20:10.52 | Beta2K | freepbx won't help with provisioning phones :) |
| 20:11.11 | [TK]D-Fender | It certainly can |
| 20:11.20 | Beta2K | Endpoint manager? |
| 20:11.25 | tm1000 | Beta2K: yes |
| 20:11.28 | bdunn | So - could I setup a DHCP server to point to a TFTP site? |
| 20:11.35 | bdunn | Endpoint Manager? |
| 20:11.36 | Beta2K | Yes |
| 20:11.48 | [TK]D-Fender | EPM |
| 20:11.49 | [TK]D-Fender | though.. it'll be for that same server |
| 20:11.49 | [TK]D-Fender | not sure you can manually point it elsewhere and jsut usse for setup |
| 20:11.57 | *** part/#asterisk rasheed (rasheed@bba402370.alshamil.net.ae) |
| 20:12.18 | [TK]D-Fender | bdunn: no need for DHCP. Just point it to a folder on whatever tech you want |
| 20:12.48 | bdunn | Do you know where I would point it? Can I do that from the web interface? |
| 20:13.00 | bdunn | (I'm not a phone guy - I'm having to cover someone) |
| 20:13.00 | Beta2K | From the BootROM |
| 20:13.14 | Beta2K | Ethernet menu I think |
| 20:13.46 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 20:13.48 | bdunn | Here's a good question - is there any way to do that from remote? I am about 350 miles away at the moment. |
| 20:18.31 | *** join/#asterisk Neptu (~Hej@c213-89-24-62.bredband.comhem.se) |
| 20:19.21 | [TK]D-Fender | bdunn: No |
| 20:19.30 | [TK]D-Fender | get someone there to push buttons for you |
| 20:21.43 | bdunn | They left until tomorrow at 1pm. Do you think I may be able to resolve this with FreePBX EPM? |
| 20:22.07 | Beta2K | If you provision them yes, assuming you can remote reboot them |
| 20:22.17 | [TK]D-Fender | If you can't change your BootROM provisioning source nothign you do matters |
| 20:22.33 | Beta2K | You should be able to change it via dhcp |
| 20:22.42 | [TK]D-Fender | not necessaryly |
| 20:22.52 | Beta2K | Yeah if their setup static you're boned :) |
| 20:23.13 | bdunn | They are all DHCP. |
| 20:23.25 | bdunn | I can remote reboot them. |
| 20:23.38 | bdunn | TFTP server? |
| 20:23.50 | Beta2K | tftp, ftp or http |
| 20:23.54 | [TK]D-Fender | depends if the PROVISIONING source is specified by DHCP Options <--- |
| 20:23.55 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 20:24.07 | [TK]D-Fender | We are not talking about them just getting their OWN IP address thre |
| 20:24.17 | [TK]D-Fender | it's a question of the PROVISIONING SERVER IP address |
| 20:24.20 | bdunn | Right - I've done this before YEARS ago and it worked well. |
| 20:24.33 | [TK]D-Fender | So depends on the phone |
| 20:25.58 | Beta2K | But code 66 should set the provisioning server ip and protocol |
| 20:26.10 | Beta2K | atleast on the 430's it does? |
| 20:26.26 | bdunn | I just realized I am going to have to have current images for the Polycom phones, which Polycom doesn't make freely available. Am I remembering that right? |
| 20:26.56 | Beta2K | No they do |
| 20:27.03 | Beta2K | Again, atleast for the 430's |
| 20:27.18 | bdunn | Have a link? |
| 20:27.41 | Beta2K | One sec |
| 20:28.01 | Beta2K | I've got a AP for these 8002's too close and it's mucking my wifi up :) |
| 20:28.44 | Beta2K | http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip650.html |
| 20:28.53 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 20:30.09 | bdunn | Thanks! I guess they changed their ways. |
| 20:31.25 | Beta2K | Fender, any idea where to start looking with these pohones that asterisk isn't answering registration requests for? |
| 20:34.31 | [TK]D-Fender | Beta2K: Not offhand... what version are you running? |
| 20:37.09 | igcewieling | start with a sip debug on the asterisk server |
| 20:37.14 | carrar | I always find things I am missing in my car |
| 20:37.22 | *** join/#asterisk Neptu (~Hej@c213-89-24-62.bredband.comhem.se) |
| 20:37.52 | [TK]D-Fender | [15:36]Beta2KAnyone around tried to get a Polycom 8002 to register? I'm seeing these incomming SIP register packets, but asterisk isn't sending anything back? http://pastebin.com/FHAbBHUX |
| 20:37.54 | [TK]D-Fender | he did |
| 20:38.07 | bdunn | I am just about ready to roll now… I have the Polycom download, setup the tftp server, setup the DHCP 66 entry. I want to have the phone point to the correct server. Which file(s) will I be looking at? |
| 20:38.12 | igcewieling | bdunn: Polycom has released their firmware for many years. |
| 20:38.41 | igcewieling | bdunn: you should read the admin guide, |
| 20:39.07 | carrar | 8002, thats their crappy spectra link stuff? |
| 20:39.17 | carrar | I've got a 8020 I use once in a while here at home |
| 20:39.51 | [TK]D-Fender | bdunn: Well you haven't told us how you expect to reconfigure them beyond the Web interface |
| 20:40.09 | [TK]D-Fender | bdunn: So with that in mind leave the STCOK files there and see if it goes into that condition by the time you're done |
| 20:40.23 | bdunn | My main goal is to clear the old settings out so I can have the Switchvox provision them. |
| 20:41.04 | igcewieling | carrar: yes. We prefer the Kirk wireless phones from Polycom |
| 20:41.24 | *** join/#asterisk mattsl (~user@c-50-142-241-34.hsd1.tn.comcast.net) |
| 20:41.30 | carrar | yeah the DECT stuff always works better thent he wifi stuff |
| 20:42.13 | igcewieling | bdunn: menu/Settings/Advanced/456/Admin Settings/Reset to Defaults and reset the device settings, then reset local config, then reset web config |
| 20:42.15 | carrar | And I'm not a big fan of having to configure a bunch of spectralinks |
| 20:42.47 | bdunn | igcewieling: Yes, but these phones aren't clearing their old settings when doing that. They are booting back up with the old settings anyway. |
| 20:43.05 | igcewieling | bdunn: exactly WHAT settings? |
| 20:43.07 | carrar | but once configured in a area with no interference they work ok, much shorter range then DECT of course. |
| 20:44.00 | bdunn | line, user, password, pointing to previous old server (different network) for server, app server, time server, etc. |
| 20:44.44 | igcewieling | you can change most of that via the phone menus |
| 20:45.01 | igcewieling | Sounds to me like the phone is using DHCP to find a local provisioning server |
| 20:45.42 | igcewieling | resetting the device settings is the important one, that clears out the FTP server/user/pass info |
| 20:45.48 | bdunn | igcewieling: Yes, but I'm 350 miles away. :-) DHCP option 66 wasn't set until I just set it. These phones are from a different network. Same model phones that are new are working just fine - provisioned by Switchvox without issue. |
| 20:46.17 | igcewieling | bdunn: if you are 450 miles away, how did you go into the phone menus and do what I suggested? |
| 20:46.42 | carrar | Like you don't have a robot you can control remotely! |
| 20:46.45 | [TK]D-Fender | the question is were they set STATIC for the provisioning source |
| 20:46.50 | [TK]D-Fender | And if you thery weren't then your changing to another server and providing stock pack should have likely blown away the firmware already on it AND the settings along-with |
| 20:46.51 | bdunn | igcewieling: Already tried it several times before. Was there last weekend. Had someone try it earlier today (no longer there until 1pm tomorrow). |
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| 20:47.09 | [TK]D-Fender | igcewieling: 350 miles away <---- |
| 20:47.25 | carrar | CAn only do this if you are 362 miles away |
| 20:47.32 | carrar | exactly |
| 20:47.33 | [TK]D-Fender | DAMMIT!!!!!!! |
| 20:47.35 | igcewieling | [TK]D-Fender: sounds like he doesn't understand what is going on and so is going on a wild goose chase. |
| 20:47.37 | bdunn | Oh… the provisioning source can be set to static. Oh damn. I was thinking if it is DHCP then it will all be DHCP. I think that's probably the issue here. |
| 20:48.00 | igcewieling | bdunn: correct, and resetting the device settings on the phone would clear that info out. |
| 20:48.02 | [TK]D-Fender | [16:23][TK]D-Fenderdepends if the PROVISIONING source is specified by DHCP Options <--- |
| 20:48.32 | igcewieling | carrar: we call them "Tech" and they are dumber than a bag of rocks. |
| 20:48.54 | bdunn | igcewieling: Okay - then I still don't quite understand. Resetting the settings on the phone using the menus isn't clearing it out. Unless it is cleared during the BOOT menu access (which makes sense now - DUH). |
| 20:49.24 | igcewieling | bdunn: there are multiple types of settings you can clean using the phone menu. Most of them do not clear out the provisioning server info |
| 20:49.52 | igcewieling | you do not need to be in the boot menu to change the provisioning server. |
| 20:50.04 | igcewieling | maybe in 10 old firmware, but nothing reasonably recent |
| 20:50.07 | bdunn | What's the easiest way to walk someone through that? |
| 20:50.12 | igcewieling | 10 YEAR old firmware. |
| 20:50.33 | [TK]D-Fender | [16:22][TK]D-FenderIf you can't change your BootROM provisioning source nothign you do matters |
| 20:50.39 | [TK]D-Fender | Ive already said it all |
| 20:50.39 | igcewieling | bdunn: (4:42:12 PM) igcewieling: bdunn: menu/Settings/Advanced/456/Admin Settings/Reset to Defaults and reset the device settings, then reset local config, then reset web config |
| 20:50.40 | igcewieling | (4:45:41 PM) igcewieling: resetting the device settings is the important one, that clears out the FTP server/user/pass info |
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| 20:51.40 | carrar | packet capture the phones request at bootup, create fake network with the hardcode provistioning server ;) |
| 20:51.50 | bdunn | Got it. Thanks a lot! |
| 20:52.42 | bdunn | While I have experts at hand - anyone have an opinion on FreePBX capabilities vs. Switchvox from Digium? |
| 20:53.10 | carrar | SwitchVox ++ |
| 20:53.19 | igcewieling | All GUIs suck. |
| 20:53.26 | carrar | except they keep changing the damn Diversion headers |
| 20:53.30 | carrar | !@#$!@(*&^~ |
| 20:53.41 | igcewieling | carrar: what do they do? |
| 20:53.51 | carrar | 5.7 they just switched to tel: |
| 20:53.54 | carrar | was sip: |
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| 20:54.05 | igcewieling | LOL! Asterisk doesn't support tel: diversion headers. |
| 20:54.38 | bdunn | I like GUIs for a lot of things, but they can really get in the way if that's all you have. |
| 20:54.42 | igcewieling | chan_sip.c: Huh? Not an RDNIS SIP header (tel:17189750600)? |
| 20:55.11 | carrar | I'm talking SIP Headers |
| 20:55.12 | igcewieling | I keep meaning to file a bug report, but I work around the issue in my AGI |
| 20:55.15 | carrar | not some console output |
| 20:55.31 | igcewieling | carrar: a SIP Divert header caused that message. |
| 20:55.56 | carrar | I don't understand your statement |
| 20:55.57 | igcewieling | we do only SIP on that box. |
| 20:56.20 | carrar | we? |
| 20:56.43 | igcewieling | carrar: I'm saying it is funny that Switchvox uses tel: Divert headers and Asterisk doesn't even support that format of a Divert header. |
| 20:56.51 | igcewieling | carrar: we, the company I work for |
| 20:56.52 | carrar | Well this is a "new feature" of 5.7 I guess |
| 20:57.23 | carrar | I guessing from the notes: "Blind transfer calls to external callers now have their caller ID set correctly." |
| 20:57.29 | carrar | it's related to that |
| 20:58.11 | carrar | Correct Diverson headers for blind transfers from queues has been a issue |
| 20:58.46 | carrar | was not expecting the uri to change like that |
| 20:58.48 | carrar | anyways |
| 20:58.55 | carrar | </soapbox> |
| 20:59.47 | carrar | but |
| 20:59.50 | carrar | other then that |
| 21:00.00 | carrar | SwitchVox is nice |
| 21:00.03 | carrar | & pretty :) |
| 21:00.27 | carrar | but also not free |
| 21:00.54 | bdunn | Definitely not free. |
| 21:01.24 | bdunn | And with this level of support that doesn't seem to be there at all for two weekends in a row, I am not sure I want to deal with another one. |
| 21:02.38 | carrar | bdunn, your on switchvox? |
| 21:02.44 | bdunn | Several |
| 21:03.08 | carrar | well you knew all that before going into it |
| 21:03.24 | carrar | but I don't mind so much |
| 21:03.28 | bdunn | Actually we purchase Gold support which is supposed to be 24/7 support from Digium. |
| 21:03.40 | bdunn | With every SV… about 12 of them. |
| 21:03.41 | carrar | been working with the software since before Digium biught it |
| 21:04.01 | carrar | I blame Tristain |
| 21:04.04 | carrar | (sp) |
| 21:04.14 | carrar | :) |
| 21:04.19 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 21:04.49 | carrar | we have every level |
| 21:05.48 | carrar | curious to know what will come after Titanium |
| 21:05.49 | bdunn | Do you happen to have a secret to getting to Digium support on the weekends? Any phone number I put in it tells me it isn't a valid phone number. |
| 21:06.38 | carrar | isn't gold m-f? |
| 21:06.57 | carrar | for calling |
| 21:07.10 | carrar | I'd have to check their levels |
| 21:07.18 | carrar | I always just open a ticket via their web |
| 21:07.25 | bdunn | They changed the names. http://www.digium.com/en/products/asterisk/support/chart |
| 21:08.19 | carrar | You need to pay a additional $27,000 for Saturday support between 2-4pm |
| 21:08.24 | carrar | ;) |
| 21:08.55 | igcewieling | carrar: that is more expensive per year than our Genband/Nextone service contract. |
| 21:08.57 | bdunn | I think I need to hire a phone guy. :-) |
| 21:09.20 | carrar | one that is on site |
| 21:09.29 | igcewieling | bdunn: why? nobody else does. 8-( |
| 21:10.21 | carrar | bdunn, so what level do you have? |
| 21:10.24 | igcewieling | "You need your IT person or IT vendor to check the network wiring." "We don't have an IT person or IT vendor!" "Well, I guess you're fucked." |
| 21:10.28 | carrar | and why can't you use the web portal? |
| 21:10.44 | bdunn | I am looking at the web portal now. I've actually never needed support before. |
| 21:10.49 | carrar | heh |
| 21:11.28 | carrar | You'll need to have a login of course already setup :) |
| 21:11.50 | carrar | and attached to the regcode |
| 21:12.15 | carrar | But I am sure the person who set this up did all that correctly for you :) |
| 21:12.21 | bdunn | Okay - so the web portal is the way to go. Thanks again. |
| 21:12.29 | carrar | yeah |
| 21:12.31 | carrar | np |
| 21:12.43 | carrar | good luck |
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| 21:35.41 | KNERD | Guss I wil try in this channel too.. |
| 21:36.08 | KNERD | am doing a script and having a problem with this line: --> menuselect/menuselect --enable format_mp3 --enable LOW_MEMORY --enable-category MENUSELECT_CORE_SOUNDS --enable-category MENUSELECT_MOH --enable-category MENUSELECT_EXTRA_SOUNDS menuselect.makeopts |
| 21:37.42 | KNERD | the options are not getting passed to menuselect |
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| 21:46.25 | igcewieling | I use = don't know if it matters. menuselect/menuselect --enable=format_mp3 --enable=cdr_mysql --disable=CORE-SOUNDS-EN-GSM --disable=MOH-OPSOUND-WAV |
| 21:53.19 | igcewieling | I also don't put menuselect.makeopts on the line, but I do run this before running menuselect: "make menuselect.makeopts" |
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| 22:09.22 | *** join/#asterisk lvlinux (~n1gg@c-50-142-165-230.hsd1.tn.comcast.net) |
| 22:10.20 | lvlinux | anybody used welltech wellgate FXO interfaces? |
| 22:11.33 | lvlinux | I'm trying to find the best FXO solution for this * installation. Only need one FXO. |
| 22:12.06 | [TK]D-Fender | what kind of call usage? |
| 22:12.16 | lvlinux | Don't know whether to go with an external gateway like the Welltech, Linksys/Sipura 3000, or a <cough> Cisco router as an FXO-SIP gateway (if you can do that :-) |
| 22:13.17 | lvlinux | what do you mean by call usage? you mean volume? |
| 22:14.13 | igcewieling | your best option for inexpensive 1 FXO may be a Linksys ATA |
| 22:14.42 | lvlinux | would that be better than a TDM card like X100p or TDM400? |
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| 22:16.32 | lvlinux | I've searched and read a lot and most info out there is just basically "I use XXX and it works fine." |
| 22:16.34 | igcewieling | there are no real X100Ps, they have not been manufactured in almost 10 years. A TDM400P may or may not be stable, older ones are not. A TDM40P will be a lot more expensive. |
| 22:17.46 | lvlinux | Ahh I wondered about that -- most say x100p compatible. |
| 22:18.13 | lvlinux | I have a couple TDM400s. One installed in another system at a different location and it seems to work fine. |
| 22:18.36 | igcewieling | I don't think I'd get an X100P for my ex-wife and I'd happily put cyanide in her coffee. |
| 22:18.55 | lvlinux | I installed another one here to test and it missed a lot of the callerid |
| 22:19.37 | lvlinux | well good I can eliminate that one lol... still left with the other options. |
| 22:19.59 | igcewieling | The issue with older TDM400Ps is that they just stop recognizing incoming calls and don't work for outgoing calls. |
| 22:20.41 | lvlinux | so you mean they have issues with age, or with newer drivers? |
| 22:20.53 | igcewieling | So if you are buying one NEW you should be OK, but if you get one from ebay or other second hand source, then you could be in for issues . |
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| 22:21.11 | igcewieling | lvlinux: hardware issue, corrected at some point. |
| 22:22.03 | lvlinux | ahhh, i see. The 3 I have are all second-hand. I think one is actually a TDM410 |
| 22:22.18 | igcewieling | We stopped using Digium cards and switched to Sangoma before the hardware/design issues were resolved, never switched back. |
| 22:23.19 | lvlinux | I have heard good things about Sangoma but never used them yet. |
| 22:24.15 | KNERD | igcewieling>: thanks I will try |
| 22:24.54 | lvlinux | So I still don't know whether an internal card or external gateway is best for just one line. |
| 22:26.37 | lvlinux | I'm interested in reliability more than features. |
| 22:27.05 | igcewieling | lvlinux: if you were interested in reliability you would go with a PRI 8-| |
| 22:27.45 | KNERD | for just a dial tone and old phoen for $10 a month |
| 22:28.32 | lvlinux | lol well yes I guess so. But it's a small office w 1 PSTN FXO line so that's all I got to work with. |
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| 22:31.01 | lvlinux | "reliability" is one of those things that is always relative... |
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| 22:32.57 | KNERD | how's this for reliability? http://olive-drab.com/images/ta312_800.jpg |
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| 22:33.26 | igcewieling | My, what big capacitors you've got! |
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| 22:33.49 | KNERD | of course! |
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| 22:35.55 | lvlinux | hehe yeah those things keep working... |
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| 22:38.00 | lvlinux | Any of you guys ever used Cisco->Asterisk stuff? (not SCCP phones) but something like I mentioned using a Cisco router as a pstn gateway? |
| 22:38.23 | igcewieling | lvlinux: long time ago, yes. |
| 22:38.54 | igcewieling | again, more expensive than a SPA ATA and gives you little advantage |
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| 22:40.56 | lvlinux | well I have several Cisco routers but no voice cards in them (I do Cisco networking), mostly lab type older routers--3600/2600/1700. Can't find a whole lot of info about doing non-Cisco stuff w them lol. |
| 22:41.22 | lvlinux | Cisco doesn't exactly provide howto's on how to use Asterisk with their stuff instead of Callmanager lol. |
| 22:42.32 | lvlinux | But I see FXO interfaces for them on eBay for about the same cost as an FXO ATA (some cheaper) so I was wondering if it would be worth it. |
| 22:43.10 | lvlinux | Can't figure out if that would be all that is needed or if the internal AIM cards would be required too or if that is just callmanger specific? |
| 22:45.12 | igcewieling | you also need DSP chips for the PSTN cards. |
| 22:46.20 | lvlinux | oh, I thought that is what the AIM modules were---no? |
| 22:47.36 | lvlinux | So basically to be able to do what I wan't I'd have to outfit a router just like it were going to be used with Callmanager? Not just buy the FXO interfaces? |
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