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| 03:42.52 | RomanDcoz | Hello Guys, I just want to block incoming request, and allow only specific IP for incoming call, Can any one suggest me how can i achieve it ? |
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| 03:45.03 | [TK]D-Fender | RomanDcoz, iptables <- |
| 03:45.09 | [TK]D-Fender | this is abasic firewall job |
| 03:45.48 | RomanDcoz | Can't it be ACL ?? |
| 03:46.14 | [TK]D-Fender | There is a differnce between IGNORING a request, and BLOCKING a request |
| 03:46.21 | [TK]D-Fender | Remeber the phrasing you use matters |
| 03:46.30 | [TK]D-Fender | as for accepting a call.. that's what your peers are for. |
| 03:46.47 | [TK]D-Fender | If you allw unathed call that make it to your server or not is entirely your choice as well |
| 03:46.57 | RomanDcoz | we have just incoming server, so just wanted to add IP and allow calls. |
| 03:47.44 | [TK]D-Fender | So firewall it so there is no chance of other IP's ever touching * at all |
| 03:48.46 | RomanDcoz | Is there any refrence link for manage IPTABLES ? |
| 03:50.14 | [TK]D-Fender | Hundreds |
| 03:50.17 | [TK]D-Fender | this is linux basics |
| 03:51.43 | RomanDcoz | Yes thanks. |
| 03:51.43 | RomanDcoz | Also i want to playback a media file for repeataly, Is there any Applications available ?? |
| 03:53.33 | [TK]D-Fender | please rephrase that |
| 03:57.04 | RomanDcoz | Rephrase means ? |
| 03:57.49 | [TK]D-Fender | fix the wrods in your question |
| 03:57.54 | [TK]D-Fender | your request was not clear |
| 03:57.58 | [TK]D-Fender | words* |
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| 03:58.36 | RomanDcoz | OK, Like if i want play an audio file repeat till the timeout, how can i achieve it ?? |
| 04:04.21 | [TK]D-Fender | Its your dialplan, make it do whatever you want. |
| 04:04.47 | [TK]D-Fender | Play the file X times. |
| 04:04.51 | [TK]D-Fender | cout how many yourself |
| 04:05.02 | [TK]D-Fender | count* |
| 04:05.31 | [TK]D-Fender | Or take the recording, and make a new one with a copy/paste of it and then time-cut it yourself |
| 04:05.49 | [TK]D-Fender | Or have an extrenal process use AMI to spy on the channel and focibly redirect it. |
| 04:05.55 | [TK]D-Fender | Take any notheod you want for it |
| 04:05.58 | [TK]D-Fender | method* |
| 04:28.22 | RomanDcoz | actullay i am using AGI for PHP , and i am doing while(1){playback file}, but after 5 min , call drop with script stop with reason 4. |
| 04:34.19 | [TK]D-Fender | Go look at the debug to see which side actually terminated |
| 05:02.13 | RomanDcoz | <PROTECTED> |
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| 13:52.09 | MacroMan | Is there anyway I can change the voice prompts that are played when performing a atxfer (setup in features.conf)? |
| 13:52.27 | Samot | Sure. |
| 13:52.41 | MacroMan | I can't find any examples or docs online to show me how |
| 13:52.42 | Samot | They're just regular prompts driven by the language of the channel. |
| 13:52.47 | Samot | What do you want to change? |
| 13:52.59 | MacroMan | I want to not play them ideally |
| 13:53.17 | MacroMan | But changing the sound to a very short bit of silence would work too |
| 13:53.21 | Samot | So you want them to hit *2 for a transfer and not get a prompt for the destination? |
| 13:53.26 | MacroMan | Correct |
| 13:53.42 | MacroMan | I'm handling the sending of the dtmf tones automatically and the prompts are pointless for me |
| 13:54.01 | Samot | Then do a blind xfer. |
| 13:54.22 | MacroMan | I don't want a blind transfer. |
| 13:54.41 | MacroMan | I'm happy with the the way it works, just want to get rid of the prompts |
| 13:54.55 | Samot | Does the user have to add digits? |
| 13:55.23 | MacroMan | Basically, my phone has a directory, so the person clicks a name, not types the number in. I still want to them the be connected to the new callee. |
| 13:56.15 | MacroMan | No, I send the atxfer dtmf code, wait 2 seconds for the voice prompt and then send the dtmf for the number automatically. |
| 13:56.18 | Samot | So why use the feature code? |
| 13:56.53 | Samot | I guess that would depend on your phones abilities. |
| 13:56.54 | MacroMan | Because it's way much easier to setup than programming atxfer into my softphone |
| 13:57.43 | MacroMan | It's literally just the audio I'm bothered by. I'm entirely happy using atxfer and the way it works |
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| 13:58.36 | MacroMan | Is there a setting in features.conf where I can change the audio file to use? |
| 13:59.00 | Samot | No. |
| 13:59.15 | MacroMan | Why did you say "sure" earlier to my question? |
| 13:59.24 | Samot | Because you can still make it silent. |
| 13:59.31 | Samot | You just have to re-record the prompt. |
| 13:59.34 | Samot | With silence. |
| 13:59.36 | MacroMan | Cool, can you point me in the right direction please? |
| 13:59.46 | MacroMan | Ah I see and overwrite the audio files |
| 13:59.57 | Samot | Correct. |
| 14:00.17 | MacroMan | Thanks. I'll roll with that. |
| 14:00.26 | Samot | And you can remove the prompt, I'm guessing by modifying the core code and recompiling. |
| 14:00.48 | MacroMan | Way beyond my ability to do that |
| 14:01.06 | MacroMan | I can compile from source, but I'm not up for modifying the source code. |
| 14:11.43 | *** join/#asterisk eaxxae (~bitopr@unaffiliated/eaxxae) |
| 14:13.04 | [TK]D-Fender | What phones are you using? |
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| 14:14.58 | *** join/#asterisk eaxxae (~bitopr@unaffiliated/eaxxae) |
| 14:17.54 | MacroMan | It's a web phone I made myself utilising webRTC |
| 14:18.32 | MacroMan | We have a very custom internal system, which it's integrated to. |
| 14:20.11 | eaxxae | can you replicate the services of grasshopper with a SIP/asterisk basic install .. or would it take a lot of custom configuration |
| 14:21.10 | MacroMan | Pretty much. I'd have to remove the stuff specific to my crm |
| 14:21.33 | MacroMan | https://imgur.com/a/xQPXI |
| 14:23.28 | MacroMan | Sorry, that wasn't aimed at me was it? |
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| 14:25.41 | MacroMan | eaxxae: Everything in asterisk takes custom configuration, so the answer is yes, but you'll need to configure it to do that. |
| 14:30.47 | MacroMan | Is there anyway I can use Application_Set in my dialplan without it blocking? |
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| 14:32.06 | [TK]D-Fender | eaxxae, Never heard of them. What do they do? |
| 14:32.40 | [TK]D-Fender | MacroMan, Almost all apps are blocking. What specific action are you using it for? |
| 14:33.17 | MacroMan | I'm using curl inside the set application for logging purposes, but it delays the start of the call. |
| 14:33.49 | Samot | [TK]D-Fender: Grasshopper is a US service. Basically hosted voice on your cell phone. |
| 14:34.06 | [TK]D-Fender | You could use System() and do it in an external process you background |
| 14:34.25 | [TK]D-Fender | Samot, What is hosted voice on a cell? |
| 14:34.40 | Samot | You know..Ring Central, Vonage.. |
| 14:34.40 | [TK]D-Fender | Actual PBX login as an app? |
| 14:34.49 | Samot | Expect they only do it via mobile apps |
| 14:34.50 | [TK]D-Fender | So jsut a softphone |
| 14:34.54 | Samot | right |
| 14:34.57 | Samot | But it's the whole service. |
| 14:34.57 | [TK]D-Fender | Ok. |
| 14:35.27 | [TK]D-Fender | eaxxae, Asterisk isn't a softphone. You can use * as a PBX for your cell to connect to and brdige to some other service if you wish |
| 14:35.35 | [TK]D-Fender | bridge |
| 14:35.48 | [TK]D-Fender | eaxxae, but you'll still need an app for your phone |
| 14:35.54 | [TK]D-Fender | For which there are many |
| 14:36.05 | Samot | Grasshopper just has their own |
| 14:36.06 | Samot | That's it |
| 14:36.11 | Samot | Like Ring Central does. |
| 14:36.13 | [TK]D-Fender | Got that. |
| 14:36.18 | Samot | What Grasshopper is doing isn't new. |
| 14:36.23 | [TK]D-Fender | nope.... |
| 14:36.25 | Samot | I'm talking to eaxxae |
| 14:36.28 | Samot | I'm talking to eaxxxae |
| 14:36.31 | Samot | whatever. |
| 14:36.48 | [TK]D-Fender | we're agreeing... so yup, whatever! |
| 14:37.00 | Samot | I meant whatever for the spelling of the name |
| 14:37.22 | eaxxae | it says asterisk is 495, but it's open source so I'm guessing that's just for support. |
| 14:37.31 | Samot | 495? |
| 14:37.36 | Samot | What does that mean? |
| 14:37.40 | eaxxae | $ |
| 14:37.53 | eaxxae | for a license, and support |
| 14:38.01 | Samot | What is? |
| 14:39.13 | eaxxae | from digium... you pay 495 for support/license. |
| 14:39.23 | Samot | Oh that's Certified Asterisk. |
| 14:39.47 | MacroMan | [TK]D-Fender: How could I make my shell call run in the background?: https://paste.ngx.cc/f4f4889b7f2eedf7 |
| 14:39.50 | Samot | Which is nothing more than Asterisk with less updates, SLAs and a few other things. |
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| 14:54.09 | [TK]D-Fender | MacroMan, call your systejm call to background like normal. |
| 14:54.15 | [TK]D-Fender | myscript.sh & |
| 14:54.18 | [TK]D-Fender | & at the end |
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| 15:03.10 | MacroMan | Ah of course. Really didn't think there. |
| 15:03.15 | MacroMan | Thansk |
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| 17:09.26 | Demon_VoIP | Where are a lot of issues for bug with asymmetric_rtp_codec. But all of them is fixed. |
| 17:09.43 | Demon_VoIP | I have asterisk 15.0.0 and asymmetric_rtp_codec=no for endpoint with different codecs... |
| 17:10.30 | Demon_VoIP | two cases: asymmetric_rtp_codec=yes and asymmetric_rtp_codec=no nothing changed in codec selection in SDP. NOTHING. But with asymmetric_rtp_codec=no - oneway audio. |
| 17:11.16 | Demon_VoIP | question: all bugs fixed in 15? |
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| 17:15.54 | Samot | asymmetric_rtp_codec=no is default. |
| 17:16.15 | Samot | Which basically means, the codec offerings have to match |
| 17:16.35 | Samot | That setting doesn't change codecs. |
| 17:16.50 | Samot | It just allows them offer/sending to not match. |
| 17:16.58 | Samot | s/them/the/ |
| 17:17.20 | Samot | Show a SIP debug of a call with one-way audio. |
| 17:17.59 | Demon_VoIP | Samot: https://qos.ruvoip.net/share/#/653ab256-8add-46f5-b250-c21043c72923 |
| 17:18.32 | Samot | That's not what I asked for at all. |
| 17:19.55 | Demon_VoIP | Samot, Tab Export, "Export TEXT" |
| 17:20.12 | Samot | No. |
| 17:20.21 | Samot | I want to see this from Asterisk. |
| 17:20.24 | Samot | Not the freaking outside. |
| 17:21.09 | Demon_VoIP | Has Asterisk (PJSIP) tools to save sip debug for two separate IP address? |
| 17:21.43 | Samot | pjsip set logger on |
| 17:22.37 | Demon_VoIP | A LOT of debugs... millions of sip dialogs from thousands endpoints. I can't filter. |
| 17:23.22 | Demon_VoIP | SIP debug is from HEP. Is it not correct? |
| 17:25.56 | Samot | WTH.. |
| 17:26.47 | Samot | One side is g279/ulaw/alaw and the other side is opus/ulaw over TLS |
| 17:27.52 | Samot | So sipper to Asterisk is ulaw and Asterisk to Zoiper is opus over TLS |
| 17:27.59 | Samot | Little transcoding going on there. |
| 17:28.12 | Demon_VoIP | 10228: endpoint: allow=alaw,g729; client: g729,alaw,ulaw |
| 17:28.12 | Demon_VoIP | demon (TLS): endpoint: allow=g729,opus,ulaw; client: opus,ulaw |
| 17:28.39 | Samot | OK |
| 17:28.46 | Demon_VoIP | 10228 calls demon. 10228 (Phoner lite) uses codecs: IN alow; OUT g729 |
| 17:28.50 | Samot | 10228 is using ulaw |
| 17:28.58 | Samot | demon is using opus |
| 17:29.13 | Samot | Make your codecs match |
| 17:29.22 | Samot | Prove this works with 100% matching codecs. |
| 17:29.35 | Samot | Then play around with orders and matching stuff. |
| 17:29.48 | Demon_VoIP | Samot, it is synthetic test, of course |
| 17:30.36 | Samot | OK |
| 17:30.45 | Samot | Doesn't change what I said. |
| 17:30.53 | Demon_VoIP | But WHY? Periodically, I lose audio. It has to asymmetric_rtp_codec=yes :( |
| 17:31.27 | Samot | Allow the sending and receiving codec to differ and not be automatically matched (default: "no") |
| 17:31.30 | Samot | ^^ That's all it does |
| 17:31.36 | Samot | Why are you messing with it? |
| 17:31.46 | *** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co) |
| 17:31.56 | Samot | Why do you need the default behaviour to be different? |
| 17:33.09 | Demon_VoIP | can "asymmetric_rtp_codec=no (default)" resuls as audio to be lost? |
| 17:33.21 | Samot | Sure. |
| 17:33.33 | Samot | That is nothing more than CODEC MATCHING |
| 17:33.49 | Samot | A ulaw/ulaw call can still loss audio. |
| 17:34.51 | Demon_VoIP | i can't use ulaw. Only alaw. |
| 17:34.59 | Demon_VoIP | i don't use... |
| 17:35.09 | Samot | I was just making a point. |
| 17:35.17 | Demon_VoIP | Is there a link where I can see the rules of selection of codecs in asterisk or PJSIP? I found only for FS. |
| 17:35.18 | Samot | same codec/same codec can have audio loss. |
| 17:35.26 | Samot | That is generally not a codec issue, that's a network issue. |
| 17:36.03 | Demon_VoIP | audio loss via NAT, packet loss... I know. Ok. Oneway audio - is it more clear? |
| 17:36.24 | Samot | One way audio is a NAT issue |
| 17:36.27 | Samot | Or a networking issue. |
| 17:37.47 | Demon_VoIP | two test: asymmetric_rtp_codec=yes and asymmetric_rtp_codec=no. In same conditions. SIP dialog - THE SAME. Different is oneway audio in case of default asymmetric_rtp_codec=no. Is it networking issue? |
| 17:38.23 | *** join/#asterisk cervajs2 (~cervajs2@178.148.broadband4.iol.cz) |
| 17:39.15 | Demon_VoIP | Is there a link where I can see the rules of selection of codecs in asterisk or PJSIP? I can't understend what must be different in case of different of asymmetric_rtp_codec values. |
| 17:39.37 | Samot | OK.. |
| 17:39.42 | Samot | Stop. |
| 17:40.05 | Samot | You have your endpoint settings on Asterisk for this device. |
| 17:40.21 | Demon_VoIP | two devices. Yes. |
| 17:40.37 | Samot | So. |
| 17:41.18 | cervajs2 | i'm reading changelog for ast15 and its looks like the binaural synthesis is merged in ast15 but there is no info about it in "new in 15". any reasons? |
| 17:41.27 | Samot | For the demon device you have an allow= setting... |
| 17:41.39 | Samot | That is where you set the allowed codecs for the endpoint. |
| 17:41.44 | Samot | In _order_ |
| 17:41.59 | Demon_VoIP | [20:28:12] <Demon_VoIP> 10228: endpoint: allow=alaw,g729; client: g729,alaw,ulaw |
| 17:41.59 | Demon_VoIP | [20:28:12] <Demon_VoIP> demon (TLS): endpoint: allow=g729,opus,ulaw; client: opus,ulaw |
| 17:42.17 | Demon_VoIP | allow=!all,g729,opus,ulaw |
| 17:42.26 | Samot | OK, why is the client offering more codecs than allowed? |
| 17:42.31 | Samot | Don't do that. |
| 17:42.42 | Samot | Why doesn't the client order match? |
| 17:43.33 | Demon_VoIP | Samot, do you offer me a way not to stumble on a bug? Or? |
| 17:43.35 | Samot | 1:34:50 PM D<Demon_VoIP> i can't use ulaw. Only alaw. <-- What is one device ulaw then? |
| 17:43.45 | Samot | What bug? |
| 17:44.01 | Demon_VoIP | Samot, it is test. once again |
| 17:44.02 | Samot | I mean first get all your codecs lined up properly |
| 17:44.34 | file | there was a bug where the asymmetric option may not change formats correctly, it's been fixed in the branch and will be in the next release |
| 17:44.44 | Demon_VoIP | Samot, i can't. It's public PBX, where any registered user can make settings he whant |
| 17:44.47 | Samot | OK. |
| 17:45.09 | Samot | So then this is a live issue. |
| 17:45.12 | Samot | Not a test. |
| 17:45.17 | Samot | Totally different. |
| 17:45.36 | Demon_VoIP | file, has 15.0.0 this bug? I don't see target releases in some issues |
| 17:45.49 | file | yes. |
| 17:46.14 | Demon_VoIP | file, this explains. Thank you. |
| 17:47.11 | Demon_VoIP | file, and.. i am not sure with correct codecs offer in some cases... Is there any url for help me? |
| 17:47.26 | file | the order of preference is the order specified in the .conf file |
| 17:47.36 | file | but the remote side can use any negotiated codec |
| 17:47.38 | Samot | ^^^ |
| 17:49.13 | Samot | But if you say you can't use ulaw... |
| 17:49.22 | Demon_VoIP | file, i don't sure.. I see offers like FS describes in it's url.. |
| 17:49.25 | Samot | Then neither your clients or the PBX should have it in the offerings. |
| 17:49.37 | file | I don't understand that statement. |
| 17:50.14 | Demon_VoIP | 10228: endpoint: allow=alaw,g729; client: g729,alaw,ulaw |
| 17:50.15 | Demon_VoIP | demon (TLS): endpoint: allow=g729,opus,ulaw; client: opus,ulaw |
| 17:50.34 | file | the preference order would be what is configured in Asterisk |
| 17:50.39 | Demon_VoIP | call 10228 -> demon. server offers to demon codecs order: g729,opus,ulaw |
| 17:50.41 | file | but like I said, the client is free to use any negotiated |
| 17:51.33 | Demon_VoIP | file, but how server know what codecs will used by client? Client answers with SDP two or more codecs... |
| 17:51.52 | file | SDP provides the negotiated codecs |
| 17:51.59 | file | it doesn't know what is in use, until it receives media |
| 17:52.13 | Demon_VoIP | ok. it's clear.. |
| 17:52.51 | Demon_VoIP | and one more question pls. Origination with two lags :) |
| 17:53.59 | Demon_VoIP | server calls legA. answer... server calls legB. answer. What codec will be used? I see slin or slin48 :( |
| 17:54.28 | file | you haven't provided any information about configuration or further details, so there is no answer to that |
| 17:54.28 | Samot | Again, it's the SDP offering. |
| 17:54.44 | Samot | You have a trunk between the two servers. |
| 17:54.50 | Samot | The trunks have codec settings. |
| 17:55.23 | Demon_VoIP | Does Local channel nothing changing in codec negotiating? |
| 17:55.44 | Samot | This is all you, file, ole buddy. |
| 17:56.26 | file | it shouldn't, but there is an active bug in 15 where codecs on outgoing weren't quite what they should be, it was partially resolved but that may be what you're seeing |
| 17:56.32 | file | like I said you'd need to provide the info |
| 17:57.47 | Demon_VoIP | file, will fix be in 15.0.1? When it might appear? |
| 17:57.55 | file | it could be fixed in 15.1.0 |
| 17:58.02 | file | 15.0.1 would only occur as a result of a security issue |
| 17:58.13 | Samot | I wouldn't lean on this bug. |
| 17:58.32 | Samot | Your codec settings are clearly not right. |
| 17:58.40 | Samot | You're allowing ulaw.. |
| 17:58.48 | Demon_VoIP | file, can you provide me issue number? |
| 17:58.48 | Samot | When it's not something you can use. |
| 17:58.57 | file | https://issues.asterisk.org/jira/browse/ASTERISK-27259 |
| 17:59.32 | Demon_VoIP | yep, i saw it. Thank you! |
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| 18:16.56 | boch | hi all, quick question, is it possible to prioritize a call group to pick up calls from? this would be for the receptionist extension which picks up calls from almost all company groups- certain groups are more important than others |
| 18:17.48 | boch | sorry, i mean when multiple extensions from different groups are ringing simultaneously |
| 18:21.33 | [TK]D-Fender | Odds are either literal order added to pickup groups, or numeric by group |
| 18:21.35 | [TK]D-Fender | Go test |
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| 18:44.37 | boch | [TK]D-Fender, great tip thank you very much |
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| 21:31.34 | cervajs2 | is in ast 15 used somehow support of rtcp feedback from lorenzo? |
| 22:00.05 | file | codec_speex uses the info if I recall, otherwise nothing as of this time |
| 22:03.39 | sibiria | does anyone know if the GSM/edge/3g/etc. mobile networks employ anything else thant he GSM codec? |
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| 22:04.52 | sibiria | it seems such a shame that G.722 really isn't used anywhere despite being so old and technically fitting within the crucial constraints |
| 22:05.25 | Chainsaw | You mean besides "HD Voice" on Polycom handsets? |
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| 22:07.44 | sibiria | i mean anything that lives on the mobile networks but not specifically on the data chanels |
| 22:08.15 | sibiria | ie. your usual mobile cell phone |
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| 23:06.38 | c|oneman | sibiria: HD-voice 722.2 AMR-WB is supported by many 3G networks |
| 23:07.09 | c|oneman | although it mostly only works for cell-to-cell calls withing the same country |
| 23:10.10 | Samot | Any carrier that peers with another carrier in a method that supports it, can do g722 between them. |
| 23:10.19 | Samot | But again, that's between them. |
| 23:10.36 | Samot | No guarantee that one side isn't sitting on a PRI that can't. |
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| 23:17.34 | sibiria | what i'm wondering plain and simply is if there's a point for us to enable g.722 for our DIDs (and outgoing calls, which are mostly to mobile phones) |
| 23:18.53 | [TK]D-Fender | <PROTECTED> |
| 23:19.06 | sibiria | or if it's just gonna be transcoded to GSM to/from the callee/caller anyway |
| 23:19.19 | sibiria | yes the providers support it, both for our DIDs and outgoing |
| 23:19.30 | Samot | The default answer is.... |
| 23:19.39 | [TK]D-Fender | How hard is it to test and see if you can hear a difference? |
| 23:19.45 | Samot | The PSTN will use ulaw/alaw, gsm, g729 |
| 23:20.13 | Samot | There may be cases where the carriers peer and can do other codecs like that |
| 23:20.19 | c|oneman | I've been asking that question for a while, I can't find any SIP provider who peers in G722 with a cellphone company in canada or the united states, although my method of research was just asking on forums |
| 23:20.32 | Samot | There is a reason for that. |
| 23:20.50 | sibiria | the entire world seems stuck with ulaw/alaw/gsm |
| 23:20.52 | Samot | There is no guarantee the final destination can support the codec. |
| 23:21.03 | Samot | Because that's how it was built. |
| 23:21.18 | Samot | Before the Interwebs |
| 23:21.24 | c|oneman | it would have to be transcoded anyway no? g722.2 is not g722 |
| 23:21.41 | c|oneman | there is no AMR-WB on asterisk is there? |
| 23:21.46 | sibiria | g.722 predates the internet at large, too |
| 23:22.33 | sibiria | (incl. the GSM networks) |
| 23:22.49 | Samot | PRI, T1 channels, etc.. |
| 23:22.57 | Samot | Are 64Kbps. |
| 23:23.06 | Samot | You know what is 64Kbps, ulaw/alaw. |
| 23:23.49 | sibiria | so's g.722 |
| 23:24.11 | Samot | Unless everyone is going to do it... |
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| 23:24.22 | Samot | Or a big enough group is going to do it, it's just not happening. |
| 23:24.30 | Samot | The investment is rather large. |
| 23:24.52 | sibiria | that was my initial point. it's in many ways a drop-in candidate, and it's been available for 30 years soon |
| 23:24.59 | sibiria | celular sound quality is awful these days |
| 23:25.10 | sibiria | there's just not an economic incentive |
| 23:25.26 | Samot | That's because in the last 30 years.. |
| 23:25.42 | Samot | Both mobile and SIP have been regarded as "fads" for their earlier years. |
| 23:25.46 | Samot | Then BOOM. |
| 23:25.48 | Samot | Explodes. |
| 23:25.58 | Samot | And nothing is out there to support a growth like that. |
| 23:25.59 | sibiria | it's a damned shame is what it is |
| 23:27.51 | Samot | Blame the small guys. |
| 23:33.11 | Samot | Like, AT&T couldn't go all Buck Rogers, 25th Century with their network and abandoned all the old stuff without having their ass handed to them. |
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| 23:34.40 | c|oneman | you'd think they could offer HD voice for their own customers reaching their own callcenter though. That would save them money in clearler customer service calls. |
| 23:35.12 | c|oneman | I think it's at least enabled for Voicemail. baby steps. |
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