| 00:03.12 | *** join/#asterisk trelane (trelane@unaffiliated/trelane) |
| 00:03.17 | trelane | is SNOM exiting the US Market? |
| 00:10.11 | drmessano | Where did you hear that? |
| 00:11.46 | drmessano | There was an article several weeks ago that they are moving to Silicon Valley |
| 00:14.52 | trelane | drmessano, I'm having trouble finding a US distributor for one of their flagship phones the D375 |
| 00:15.00 | trelane | I can pay in pounds |
| 00:15.01 | trelane | or euros |
| 00:15.07 | trelane | or go #$%^ myself |
| 00:16.01 | trelane | trust me I do not want them to leave, their phones are the best |
| 00:18.30 | drmessano | IDK.. I dont see any at vendors I use |
| 00:18.37 | drmessano | How new is it? |
| 00:18.57 | trelane | I don't know, I've been out of the asterisk game for 3 years, just getting back in. I always used SNOM because they don't break. |
| 00:19.56 | trelane | and honestly I'm a bit ashamed of what I'd do to have that on my desk (I assume SNOM is still awesome) |
| 00:20.05 | drmessano | If the phone is new, maybe no one here is distributing it yet. The little documentation I found is from the end of April |
| 00:20.59 | drmessano | Contact someone like Telephony Depot and ask them |
| 00:21.14 | drmessano | They can check with Snom or may even know already |
| 00:21.20 | trelane | ok cool |
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| 00:42.39 | Xpress_ | Does mohsuggest work in pjsip? I have the following in pjsip.endpoint_custom_post.conf http://pastebin.com/031aeCW8 but the moh is still default when using the 101 endpoint. Using asterisk 13.9.1 works fine in sip |
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| 00:51.06 | drmessano | Xpress_: moh_suggest |
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| 01:17.26 | Xpress_ | drmessano, thanks works perfect |
| 01:32.10 | trelane | drmessano, looking for a phone for my desk, is SNOM still the best option? My polycom 501 appears to be toast. |
| 01:45.15 | drmessano | trelane: There is no BEST option.. just what works for you |
| 01:45.39 | drmessano | Im not sure i've ever heard Snom referred to as a best option |
| 01:45.57 | drmessano | They all have strengths and weaknesses |
| 01:47.08 | Xpress_ | drmessano, this doesn't work the way I expected it to, say you have a queue with MOH of default and then a moh_suggest of somethingelse on the agent it works, if you have the queue on something other than default then when the agent puts someone on hold it doesn't work and plays what ever was the moh class on the queue |
| 01:56.34 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:56.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 02:53.37 | Xpress_ | any ideas on this drmessano |
| 03:01.38 | wyoung | do I need hylafax and IAXFax installed to receive T.38 faxes from ITSP and email as a PDF attachment? Or can Asterisk do this out of the box? |
| 03:02.24 | wyoung | also, is there a test faxing service available anywhere out there? |
| 03:07.39 | trelane | very much wants to murder this polycom IP501 for being so brutally SLOW |
| 03:08.36 | wyoung | hands trelane a snom |
| 03:08.43 | trelane | wyoung, don't I wish |
| 03:09.09 | trelane | wyoung, what's the verdict on current snom models btw? 7xx series any good? better off with the 375? |
| 03:09.18 | trelane | I'm jsut getting back into PBX work after a 3 year hiatus |
| 03:09.27 | wyoung | I like them, the D7xx series are better (1GBps switch) |
| 03:09.27 | shido6 | :) |
| 03:09.42 | wyoung | 7xx are 100Mbit |
| 03:10.01 | trelane | D7xx then whatever's current |
| 03:10.12 | wyoung | and stay away from the UC models :) unless you want to use MS lync |
| 03:10.23 | trelane | nothing nice to say about MS Lync |
| 03:10.28 | trelane | it should also be murdered. |
| 03:10.29 | wyoung | never used it |
| 03:10.37 | wyoung | don't intend to |
| 03:10.42 | trelane | wyoung, correct |
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| 03:13.23 | trelane | wyoung, what snom do you/would you have on your desk if money isn't an issue? |
| 03:18.08 | *** join/#asterisk u0m3 (~u0m3@86.120.82.37) |
| 03:32.41 | igcewieling | trelane: I've noticed if the phone can't upload logs or recent calls lists it can cause the phone to lag badly |
| 03:32.49 | igcewieling | Specifically Polycoms. |
| 03:33.06 | trelane | igcewieling, this issue is the 501 is mostly shit, and it takes 5 minutes to reboot, and only applies settings if you reboot |
| 03:34.26 | igcewieling | trelane: Hmm..latest firmware is 3.18, that might be old enough to not have the boot speed enhancements. Are you running split or combined images? |
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| 03:35.03 | trelane | split iirc, but it doesn't matter it's 3 minutes and another to get a web interface |
| 03:35.21 | igcewieling | *nod* the web interface is useless |
| 03:36.55 | igcewieling | We have....a zillion Polycoms, very very few of the SPIP 501 vintage though. |
| 03:40.02 | trelane | igcewieling, best I can tel the entire interface is useless |
| 03:40.45 | igcewieling | They massive improved the interface in the 4.x series firmware, and again in the 5.x series |
| 03:41.01 | trelane | which this phone can't use |
| 03:41.15 | igcewieling | Get out of the stone age. 8-| |
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| 03:44.04 | trelane | igcewieling, I think I want the new Snom d375, but am waiting for it to make it to US distributors who are currently charging scalper prices |
| 03:44.14 | trelane | hence why I'm trying to resurrect this piece of shit |
| 03:45.24 | igcewieling | Did you hear Polycom was acquired by Mitel? |
| 03:47.12 | trelane | no, I have been out of the PBX world for ~4 years |
| 03:47.37 | trelane | back in my day sonny we had to type our dialplans :-p |
| 03:47.51 | igcewieling | heh, I started with Asterisk in late 2001. |
| 03:48.06 | trelane | igcewieling, I filed the first security defect against 1.0 two days after it was released |
| 03:48.14 | igcewieling | cool! |
| 03:48.29 | trelane | hrm every time I run nessus against the asterisk box it goes splat |
| 03:48.31 | trelane | lets figure out why |
| 03:48.50 | igcewieling | gads, Mitel also bought Astra |
| 03:48.53 | trelane | chan_skinny didn't handle garbage well |
| 03:49.06 | trelane | so if you netcat /bin/ls to it, asterisk crashed |
| 03:52.00 | wyoung | trelane: I use D715, cost me around $130AUD |
| 03:55.03 | trelane | I kind of like how the 3xx series is laid out better |
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| 04:44.15 | overyander | anyone available to help diag a fax issue? |
| 04:45.50 | WIMPy | ~ask |
| 04:45.50 | 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. |
| 04:49.24 | overyander | I'll provide an abundance of log details in just a moment. the only reason i was asking is that many people either don't know or don't want to help with faxing in * |
| 04:50.06 | WIMPy | The only thing I know is that noone will help if there's nothign to help with. |
| 04:51.25 | overyander | true, i can't argue with that. :) |
| 04:59.39 | overyander | I have a fax machine hooked to a sip ATA and configured as per http://www.asteriskpbxsystems.com/cisco-spa122-fax.html I am able to receive faxes fine, however, i cannot send. here's the sip debug on the sending server http://pastebin.com/scsBrJdG I have also done a packet capture, which in wireshark, shows two separate calls when in fact the fax is only dialing once. i think this may have something to do with my sip providers method/policy on |
| 04:59.39 | overyander | <PROTECTED> |
| 04:59.40 | overyander | <PROTECTED> |
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| 05:12.31 | drmessano | Your pastes all expired.. |
| 05:13.57 | overyander | drmessano just a sec |
| 05:16.53 | overyander | sip debug http://pastebin.com/ii4J1HUU leg 1 http://pastebin.com/8sEeVPdR leg 2 http://pastebin.com/k42tQqtd |
| 05:17.00 | overyander | drmessano ^^^ |
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| 05:24.28 | drmessano | Pastebin your Flowroute peer config |
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| 05:28.33 | overyander | drmessano it's in a db |
| 05:29.58 | overyander | drmessano here's the row from the db http://pastebin.com/1fgApWAx |
| 05:31.58 | overyander | drmessano here's the fax peer too if that helps http://pastebin.com/XfPr5bkH |
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| 05:51.20 | overyander | drmessano any suggestions or ideas? |
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| 06:12.43 | DanQuinney | Anyone around for a quick Asterisk timings question? I have a Queue that every 15 seconds plays an audio file however during testing it appears that the audio is played ~ 5 seconds after it should; http://pastebin.com/raw/JXQQt7bn |
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| 07:29.20 | fling | Where to get a DID for receiving a single sms? :P |
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| 13:19.08 | x-fak | hi |
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| 14:23.56 | bounceman | Hi, how can I tell asterisk to send PAI on outgoing calls? I've set sendrpid = pai but it does not show. |
| 14:24.39 | [TK]D-Fender | Show us |
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| 14:25.52 | bounceman | What do you want to see? |
| 14:26.56 | [TK]D-Fender | The config & the call |
| 14:28.04 | bounceman | http://pastebin.com/wSbX55W5 outgoing invite |
| 14:28.29 | [TK]D-Fender | ... |
| 14:28.40 | [TK]D-Fender | proper full call with proof of what we're loking at |
| 14:29.05 | bounceman | Full call? You meaning that PAI would be added not in the INVITE but maybe the SESSION IN PROGRESS or ACK? |
| 14:29.07 | bounceman | I do not follow you |
| 14:29.25 | [TK]D-Fender | proper full call with proof of what we're loking at |
| 14:29.46 | bounceman | Is the PAI being added in the invite or not? If it's not sure I can extract it |
| 14:29.59 | [TK]D-Fender | I want to see the full path that call goes through Dialplan and all |
| 14:30.05 | [TK]D-Fender | and do NOT fuck with the evidence |
| 14:30.15 | bounceman | You want me to post my IPs as well? |
| 14:30.23 | [TK]D-Fender | And I want to see the matching config. Mask ONLY the secret |
| 14:30.33 | [TK]D-Fender | I don't care if you PW & PM it |
| 14:30.55 | [TK]D-Fender | But I don't waste my time when the evidence can't be trusted |
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| 14:34.58 | shido6 | bounceman: please pastebin the relevant sip config and dialplan config for the call. |
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| 14:35.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:35.50 | bounceman | Realtime asterisk, really a bugger to extract :p |
| 14:36.09 | shido6 | put on your big boy pants and do what is necessary for assistance. |
| 14:36.27 | shido6 | grown man pants? |
| 14:38.11 | subvhome | Ok.. so I ditched AsteriskNOW and used the FreePBX iso from the site and installed it. Its currently asking me if i want SIPStation. This is my first time working with anything asterisk. I'm trying to get my home office up with 4 softphones. What are things that I need. I currently have 1 line that I want to work with. |
| 14:39.34 | [TK]D-Fender | clarify "1 line that I want to work with" |
| 14:40.15 | ||cw | [TK]D-Fender: from yesterday's convo, 1 incoming line |
| 14:40.23 | [TK]D-Fender | what KIND of line? |
| 14:40.35 | [TK]D-Fender | "incoming" is not a "kind" |
| 14:40.48 | ||cw | subvhome: sipstation is a provider, do you want to use them? |
| 14:41.12 | ||cw | POTS was the assumption |
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| 14:46.56 | subvhome | ||cw: with SIPStation... i can call outside lines and outside lines can call in |
| 14:46.57 | subvhome | ?? |
| 14:47.10 | [TK]D-Fender | yes |
| 14:47.17 | ||cw | that's what the marketing says |
| 14:47.18 | [TK]D-Fender | ~itsp |
| 14:47.18 | 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. |
| 14:47.20 | [TK]D-Fender | ^ |
| 14:47.44 | [TK]D-Fender | They are one of hundreds or providers |
| 14:47.46 | [TK]D-Fender | of |
| 14:47.58 | x-fak | what happens if my "Regional settings" into my SPA112 are not correctly setup, the tones...? |
| 14:48.04 | subvhome | I do apologize in advance. I'm not savvy with VoIP.. but I am a firm believer of RTFM |
| 14:48.29 | subvhome | Ok.. so I can just ditch POTS/PTSN and use the SIPStation |
| 14:48.34 | [TK]D-Fender | You could |
| 14:48.36 | subvhome | or similar |
| 14:48.40 | [TK]D-Fender | Or any other similar provider |
| 14:49.12 | subvhome | I will read on it.. Any recommendations on the provider? |
| 14:49.43 | [TK]D-Fender | Got a POTS line you're stuck with and want to use? You can get a hardware interface for it. Just want to use an ISTP? Sure. Just want to use your system as an in-home intercom? Sure. |
| 14:49.50 | [TK]D-Fender | Where are you located? |
| 14:49.56 | subvhome | NY |
| 14:50.06 | subvhome | New York |
| 14:50.11 | [TK]D-Fender | flowroute ,vitelity , voip.ms |
| 14:50.35 | subvhome | I will look into it thanks. |
| 14:55.36 | x-fak | do i need to change my "Dial Tones" into my ATA setup? |
| 14:56.49 | [TK]D-Fender | x-fak, Do you? You should be telling us your need, not asking if we think you have one for a given feature... |
| 14:58.36 | x-fak | [TK]D-Fender i'm currently setting up my CISCO SPA112 ATA, the default settings for the tones (regional into web interface) are for US anlog phone and i'm European |
| 14:58.55 | [TK]D-Fender | Good odds then |
| 15:03.25 | x-fak | ok |
| 15:03.39 | x-fak | my setup is currently working |
| 15:04.11 | x-fak | but the ringing is a bit strange, not like with Analog phone, it's irregular |
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| 15:40.13 | stefan27 | has anyone tested how having lots of local channels impacts performance? That is, if the dialplan has lots of Dial(Local/...) so that any call from SIP A to SIP B passes through 2 sip channels and 10 local channels SIP_A -> LocalX;1 -> LocalX;2 -> LocalY;1 -> ... -> SIP_B resulting on 6 times higher average "core show channels" count than just SIP to SIP... Will this cause negligable extra |
| 15:40.13 | stefan27 | cpu load? |
| 15:42.32 | rmudgett | stefan27: If you allow those local channels to optimize out then there won't be any difference in CPU load |
| 15:46.23 | rmudgett | I think leaving those local channels in the chain will result in higher CPU load as each local channel pair in the chain has to handle every frame passed between the endpoints. Likely not a negligible difference. |
| 15:46.53 | rmudgett | Also those local channels will be continuously trying to optimize themselves out of the chain. |
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| 17:06.48 | noko | There is any documentation how to use OOH323 for outgoing calls? |
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| 17:19.49 | DanQuinney | <PROTECTED> |
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| 17:28.14 | [TK]D-Fender | noko, same concept as just about any other channel driver |
| 17:28.37 | [TK]D-Fender | Dial(ooh323/peer/number", etc |
| 17:29.27 | [TK]D-Fender | DanQuinney, that shouldn't be played every 15 seconds, it should be 15 seconds between playing. |
| 17:29.45 | [TK]D-Fender | time between, not time from start to next start |
| 17:29.57 | DanQuinney | ah, that makes sense |
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| 17:30.17 | DanQuinney | tbf I didn't check the times between plays |
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| 20:33.59 | syadnom | anyone know when this issue might be resolved? https://issues.asterisk.org/jira/browse/ASTERISK-26021 |
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| 20:37.15 | newtonr | syadnom, file is working on it. I don't know if there is an ETA yet |
| 20:37.27 | file | hmmm? |
| 20:37.36 | file | yes, working on it |
| 20:37.46 | syadnom | this is holding my back on asterisk 11.22.0 :( |
| 20:38.11 | ||cw | that pages says it went from open to in progress like right now, so... |
| 20:38.32 | syadnom | ||cw, ah, when I pasted this I'm sure it just said open :) |
| 20:38.34 | file | yeah, I forgot to assign it to myself |
| 20:38.36 | ||cw | or the time listed not localized to me? |
| 20:38.38 | newtonr | ||cw, I just updated it to reflect reality |
| 20:39.00 | ||cw | k, was just suspicious |
| 20:39.04 | newtonr | lol |
| 20:39.22 | syadnom | yeah, I got hooked on siren7 and now I can't give it up :/ |
| 20:39.29 | newtonr | we are human, we forget to update the status on issues |
| 20:39.54 | syadnom | you guys are awesome, no worries. |
| 20:40.20 | newtonr | syadnom, I was hooked on phonics once. |
| 20:41.03 | syadnom | i hated phonics lol, it didn't get it's hooks into me |
| 20:41.42 | syadnom | hear it was a gateway drug to grammar and other heavier language addictions |
| 20:42.13 | rmudgett | s/addictions/dictions/ |
| 20:42.18 | newtonr | yeah i'm all the way into philosophy now - living on the edge |
| 20:42.39 | syadnom | lol rmudgett |
| 20:42.51 | newtonr | rmudgett, perfect |
| 20:47.06 | igcewieling | hugs 11.21.2 |
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| 22:43.38 | jjulian | hey i just recently added a gigaset dect receiver to my asterisk. This device seems like its a turning signal. Sometimes it works sometimes not. Its just an internal call. Anyone got a idea? Its always one or two signals and then busy tone |
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| 22:55.43 | jjulian_ | hey i just recently added a gigaset dect receiver to my asterisk and network. Calls from DECT device to internal or external works fine. Calls to the DECT devices do work one time and then i get always one free tone and then a busy tone. any ideas? |
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