| 00:00.44 | volga629 | Maybe somebody so behaviour where delivery voice massage to INBOX, and defined folder in voicemail.conf |
| 00:00.56 | volga629 | not |
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| 00:06.48 | joobie | hey guys.. when i use dial() it soudns like a US ringing tone before the person answers |
| 00:06.50 | joobie | how can i change this ? |
| 00:07.21 | WIMPy | Do you use option r to Dial? |
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| 00:07.31 | WIMPy | Where do you dial out? |
| 00:07.58 | joobie | no |
| 00:08.00 | joobie | I just use dial() |
| 00:08.14 | joobie | just diallinga nother sip phone for example,. registered on the same asterisk box |
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| 00:08.37 | WIMPy | In that case it's your phone generating the sounds. |
| 00:08.51 | joobie | ahh |
| 00:08.54 | joobie | i have a polycom 321 |
| 00:08.58 | joobie | any idea where this setting is? |
| 00:09.22 | WIMPy | Nope. They didn't cross my way so far. |
| 00:09.32 | joobie | ok thanks WIMPy |
| 00:09.39 | joobie | for knowledge sake, how do you know it's the phone's config? |
| 00:10.04 | WIMPy | If you dial another local sip phone there's nothing to generate sounds. |
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| 00:10.12 | joobie | ahh |
| 00:10.21 | WIMPy | And the usual sip way is to generate them locally. |
| 00:10.36 | joobie | you're right WIMPy |
| 00:10.45 | WIMPy | But jou can have early media. e.g. if you dial out to the PSTN or use option r. |
| 00:10.46 | joobie | i jsut dialed via a sip provider and got the proper tone |
| 00:11.58 | joobie | thanks WIMPy |
| 00:12.31 | WIMPy | You can play around with early media and progressinband. |
| 00:13.18 | joobie | i had a read into early media on the voip-info page |
| 00:13.42 | joobie | it mentions how it can sometimes throw off the busy tones (ie. it will ring .. and then if it gets busy it will change the tone to busy) |
| 00:13.50 | joobie | for this reason, going to try implement on the phone itself to avoid it |
| 00:14.57 | WIMPy | Yes, that's likely if you use option r. |
| 00:15.12 | WIMPy | However it can happen in the PSTn as well. |
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| 00:16.24 | joobie | if the carrier is using the equivilent of our dial() with the r option? |
| 00:16.58 | WIMPy | No, if the called party rejects a call after it has started ringing. Typically on a waiting call. |
| 00:17.06 | joobie | ahh good point |
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| 03:40.45 | patb | looking for help trying to get linksys spa connected to asterisk |
| 03:41.28 | patb | basilcy i am looking to get a web enabled phone to connect to asterisk and the linksys spa to connect to my home phone |
| 03:41.32 | patb | it possible? |
| 03:41.54 | patb | got gui up just cant get authentication |
| 03:42.03 | patb | on ubuntu btw |
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| 04:19.18 | Node_403 | hello |
| 04:19.56 | Node_403 | I need to use 32 GSM-VOIP cannels of my gateway to run with 300kbps Bandwidth |
| 04:20.06 | Node_403 | I need to use 32 GSM-VOIP cannels of my gateway to run with 300kbps Bandwidth |
| 04:21.52 | Node_403 | I need to use 32 GSM-VOIP cannels of my gateway to run with 300kbps Bandwidth |
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| 04:25.40 | Node_403 | I need to use 32 GSM-VOIP cannels of my gateway to run with 300kbps Bandwidth |
| 04:26.13 | Node_403 | any one can help? |
| 04:26.22 | drmessano | Why do you keep repeating? |
| 04:26.42 | drmessano | You PM'ed me, never answered, and now you're spamming in here |
| 04:26.45 | Node_403 | i need it very much |
| 04:26.48 | drmessano | Knock it off |
| 04:27.12 | Node_403 | can i PM u? |
| 04:27.38 | drmessano | NO |
| 04:27.56 | drmessano | Dont ask me and then do it before I even answer |
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| 04:28.38 | Node_403 | i think u can hel |
| 04:28.40 | Node_403 | help |
| 04:28.42 | drmessano | I will help you for 1 million US dollars consulting fee. Nothing less. Payable in US treasury bonds |
| 04:29.07 | drmessano | It might be cheaper to get more bandwidth |
| 04:29.08 | Node_403 | hmm |
| 04:29.15 | Node_403 | i can pay thousand |
| 04:29.20 | Node_403 | :) |
| 04:29.58 | drmessano | We discussed this earlier.. you need IAX2 in trunk "mode" using G729 to get 32 channels in 300k of bandwidht |
| 04:30.03 | drmessano | What is the problem? |
| 04:30.20 | Node_403 | VPN needed |
| 04:30.46 | Node_403 | im behind firewall and I dont have a public IP |
| 04:31.18 | drmessano | ROFL |
| 04:31.23 | drmessano | You can forget 32 channels then |
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| 04:31.53 | drmessano | Kiss half of that goodbye with VPN overhead |
| 04:32.13 | Node_403 | then what can be done? |
| 04:32.20 | drmessano | You need more bandwidth |
| 04:32.26 | drmessano | 300k isnt enough |
| 04:33.26 | Node_403 | I am using Wimax and its their besk package. 2mbps downlink and 300-320kbps uplink |
| 04:33.49 | drmessano | Then you are, as we say, "shit out of luck" |
| 04:34.03 | drmessano | Its just not possible |
| 04:34.14 | drmessano | 16 channels, MAYBE, with the VPN.. |
| 04:34.26 | Node_403 | It is happening in my country |
| 04:34.36 | drmessano | and they wont be decent calls, because TCP encapsulation of UDP is horrible |
| 04:34.46 | drmessano | Listen |
| 04:35.13 | Node_403 | frame sizing? |
| 04:36.06 | drmessano | You can tell me that dogs are flying over the fucking moon... The MATH doesnt support your argument. We already determined that 32 channels of G729 with IAX2 TRUNKING is going to consume about 250k of that 300k. Adding 128k of VPN overhead is going to DEMOLISH that number down to 16 CHANNELS or less |
| 04:36.50 | drmessano | Theres also additional Ethernet overhead and you need to add in varying RF conditions and latency |
| 04:37.13 | drmessano | Even the 32 channels without VPN was going to be JUST BARELY making it |
| 04:37.38 | Node_403 | any other wat arround? |
| 04:37.45 | Node_403 | any other way arround? |
| 04:37.56 | drmessano | You cant adjust the frame sizing down any further. Even using Speex at 6.4 wont get you but another channel or so |
| 04:38.04 | drmessano | Yes, use a different connection |
| 04:38.17 | drmessano | Your connection does not support the requirement. Period |
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| 04:39.46 | Node_403 | i have two asterisk pc in both end |
| 04:40.36 | drmessano | I understand that |
| 04:40.47 | drmessano | The PIPE IS TOO SMALL |
| 04:41.25 | Node_403 | if frame size is 40ms? |
| 04:41.27 | drmessano | VPN + G729 + IAX2 Trunking over 300k is going to give you maybe 15-16 usable concurrent calls |
| 04:41.46 | kaldemar | why is VPN needed? |
| 04:42.10 | Node_403 | behind firewall so |
| 04:42.32 | kaldemar | that does not equal to a VPN requirement |
| 04:43.05 | Node_403 | voip packet discard |
| 04:43.18 | drmessano | VoIP is illegal in his country |
| 04:43.21 | drmessano | So its blockedc |
| 04:43.32 | drmessano | So he needs a VPN.. cant be used over public internet |
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| 06:08.47 | greenwolf | is anyone around im experiencing funny problems |
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| 06:15.45 | kaldemar | ~ask |
| 06:15.45 | 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. |
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| 06:23.06 | volga629 | How I can set password on 011 dialing or some authentication ???? |
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| 06:26.25 | kaldemar | volga629: core show application Authenticate |
| 06:28.57 | volga629 | Or I see |
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| 06:34.41 | bulkorok | hi |
| 06:39.27 | volga629 | Some think like this exten => _1ZXXNXXXXXX Authenticate(/tmp/password_file ? |
| 06:39.32 | volga629 | ) |
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| 06:41.46 | kaldemar | volga629: your syntax is screwed up, but something like that. |
| 06:45.04 | volga629 | yes syntax not right just example |
| 06:47.35 | volga629 | I wonder if I can put macro to catch 011 and it will ask last name run to radius which hooked to secure ID and it will prompt for token if it company authorized token grand access to call |
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| 06:49.49 | volga629 | kaldemar: thank point me for right info |
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| 06:53.05 | Maxus2 | Hi Asterisk people, Is there a way to access the iax rtcp qos details for a call from the extensions.conf? |
| 06:53.27 | Maxus2 | i can access sip using: ${CHANNEL(rtpqos,audio,local_ssrc)} |
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| 06:53.52 | Maxus2 | which i do in my hang up, but iax doesn't appear to support any of those parameters. |
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| 06:58.10 | kaldemar | Maxus2: IAX does not use RTP, hence there are no RTCP parameters for it. |
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| 07:02.27 | Maxus2 | I notice int he IAX source ther are some jitter parameters being set |
| 07:02.36 | Maxus2 | how do i access those from the dial? |
| 07:02.45 | Maxus2 | bassically we want to log call quality |
| 07:03.34 | Maxus2 | acess the details in "iax2 shoe netstat" so we can store them in cdr? |
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| 07:10.12 | kaldemar | Maxus2: no pretty way of getting the stats in dialplan. maybe something ugly like Noop(${SHELL(asterisk -rx "iax2 show netstats | awk '/${CHANNEL(name)}/ { print $5 }'")}) for lost frames. |
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| 07:12.17 | Maxus2 | Okay thanks! |
| 07:12.29 | Maxus2 | might have to do some custom dev work :( |
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| 07:31.50 | jacc0 | hi all! good morning! |
| 07:32.08 | jacc0 | I'm looking for a good way to send an email from dialplan |
| 07:32.13 | jacc0 | anyone? |
| 07:32.16 | jacc0 | :) |
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| 07:44.38 | rjvvliet | jacc0: i think the system() command can help you with that. |
| 07:45.05 | jacc0 | ;) |
| 07:45.17 | jacc0 | guess I will go for agi/php then |
| 07:45.37 | rjvvliet | jacc0: Yep, i think it gives you more control. |
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| 07:47.10 | evangelion | hello |
| 07:47.36 | evangelion | how can i debug *LONG* queues shown by "core show taskprocessors" ? |
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| 07:59.27 | apurvtwr | hi |
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| 08:03.01 | apurvtwr | I am trying to create a dialplan where on the event of an inbound call at an extension, asterisk makes an outbound SIP call to a server and bridges the two channels. I want to know if it is possible to do this using asterisk. if so, how? |
| 08:04.19 | jacc0 | isn't that what the dial() command does? |
| 08:06.05 | jacc0 | @apurvtwr: in your question it sounds like you want make a 'normal' call. is that what you want? then use the dial() command |
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| 08:09.24 | apurvtwr | jacc0, thanks! I am very new to asterisk. can I set the IP and port do dial too in this command (I am looking at the document here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial) |
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| 08:10.28 | rjvvliet | apurvtwr: if you are that new to asterisk i suggest you start with this book http://www.asteriskdocs.org/ |
| 08:12.41 | apurvtwr | thanks rjvvliet . I will go through that. |
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| 08:50.02 | jacobw | hi, i'm using a SIP trunk, OUT_2 = SIP/mytrunk in extensions_additional.conf (i'm using FreePBX to manage the files) with OUTCID_2 = +44mynumber |
| 08:50.17 | kaldemar | ~freepbx |
| 08:50.17 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 08:52.15 | jacobw | in my INVITE my contact header is set to %2b44mynumber, instead of +44mynumber |
| 08:53.00 | jacobw | i think this is still an asterisk question, freepbx only wirtes th econfiguration files, the SIP message contruction and presumably this substiton is done by asterisk |
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| 08:54.31 | jacobw | unless a freepbx configuration is doing the substitution in some macro applied after [globals] |
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| 08:59.03 | kaldemar | jacobw: asterisk does not escape a + set into caller id. |
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| 09:03.26 | jacobw | ok, thanks, i'll ask in #freepbx also |
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| 09:04.05 | jacobw | kaldemar: i'm finding a lot of discussion round uri_encode on the review board for 1.8, unfortunately i'm using 1.6 |
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| 09:29.28 | timahvo1 | HI guys, after running make menuselect is there a file that is created that can be copied to multiple installs to create multiple identical installs ? |
| 09:30.22 | timahvo1 | such that it eliminates the need to run make menuselect over again in a different install |
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| 09:40.04 | aberrios | Can anyone recommend a rugged type dect phone for use with * |
| 09:47.17 | RiceCracker | n00b question: would there be a spcific router or method to splitting the use of a t1 to two asterisk boxes...i.e. one box is the production, the other testing box. |
| 09:49.05 | RiceCracker | simply asked: how can i split a t1 line for two asterisk servers? (if possible) |
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| 10:04.55 | aberrios | RiceCracker: I did this with a Sangoma Vegastream gateway. But there is probably a cheaper option. |
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| 10:21.07 | *** join/#asterisk Rolandow (~rolandow@92.68.81.83) |
| 10:22.01 | Rolandow | goodafternoon .. i'm having issues with an ATA .. it can dial out to my extensions.. but i cannot receive calls. i see a retransmission timeout |
| 10:22.06 | Rolandow | any ideas? :) |
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| 10:26.38 | jol | Is there someway to stop the current AGI command, making it end early? Such as STREAM FILE on a long sound file, and you want to stop the playback and let the AGI continue. |
| 10:27.37 | jol | AMI Redirect would be possible I guess, but that would need then to relaunch the AGI, not just continue execution after current AGI command |
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| 10:36.21 | kaldemar | Rolandow: show the output of "sip show peers" and a CLI output of a call with sip debug enabled. |
| 10:38.07 | F|shie | jol: what exactly do you want to do? end AGI or let AGI continuee |
| 10:44.01 | jol | F|shie, I want to let the AGI continue, basically letting the STREAM FILE or similiar action return without finishing completly |
| 10:47.03 | F|shie | jol: what lib you are using for AGI? |
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| 11:02.23 | Get_The_Fish | anyone know if there is going to be a yum repo for Centos 6 anytime soon? |
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| 11:03.32 | jacc0 | how do I get the insertID from a ODBC_INSERTAUTONUMROW() ? |
| 11:04.08 | Rolandow | kaldemar: what should i look for in the sip debug info? it's a lot of information :) |
| 11:04.13 | Get_The_Fish | I think that you can map the field |
| 11:04.29 | Dovid | hi all |
| 11:04.42 | jacc0 | how? |
| 11:06.32 | jol | F|shie, No lib, raw line by line (over TCP actually. "FastAGI") |
| 11:07.58 | Get_The_Fish | jacc0 im looking |
| 11:08.47 | kaldemar | Rolandow: address where asterisk sends messages. you can pastebin the trace and let someone else tell you what is wrong. |
| 11:08.50 | kaldemar | ~pb |
| 11:08.51 | 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. |
| 11:10.20 | Rolandow | http://pastebin.com/8BSnU8Dw - this is the sip debug of the retransmission request |
| 11:10.45 | Rolandow | kaldemar: but i think it's being sent to the correct IP |
| 11:11.02 | Rolandow | kaldemar: 172.16.0.34 is the ip of my ata |
| 11:11.12 | Rolandow | so that's ok |
| 11:12.14 | Get_The_Fish | https://wiki.asterisk.org/wiki/display/AST/Function_ODBC |
| 11:14.45 | Rolandow | could the SLIC setting have something to do with it? i'm not sure what it's value should be. |
| 11:15.28 | Get_The_Fish | jacc0 this might help to start. http://www.savelono.com/linux/asterisk-post/how-to-query-a-database-from-mysql-with-asterisk-16-odbc.html |
| 11:15.29 | Rolandow | i did set it to europe something.. now i will try standard (900 ohms) .. |
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| 11:21.07 | kaldemar | Rolandow: the ATA is not in the same subnet as asterisk. is there a NAT between them? |
| 11:26.54 | jacc0 | thnx, will have a look |
| 11:28.35 | Rolandow | kaldemar: our subnet is 255.255.0.0 .. there is no NAT between them |
| 11:29.03 | Rolandow | kaldemar: but perhaps the ata thinks so by mistake ? :S |
| 11:31.16 | jacc0 | Get_The_Fish: there is noly info onhow to use ODBC in asterisk. I know how that works. But I'm looking for a fail prrof way of getting the lastinsertid(). Becasue when i query 'select lastinsertid()' multiple times at about the same time they all return the same value when I've inserted multiple rows :S |
| 11:31.43 | Rolandow | kaldemar: also I am able to do an outbound call to my cellphone, so i think the ata must 'see' the asterisk server, right? |
| 11:41.37 | Rolandow | kaldemar: hmm .. this is funny .. when i call the sip account directly from my "missed calls" dir, it works fine.. |
| 11:41.47 | Rolandow | kaldemar: when asterisk is between them, it's not so fine.. |
| 11:42.10 | jacc0 | then you are dialing the wrong number :) |
| 11:42.30 | Rolandow | jacc0: hehe, no .. because in the logs i see the same sip account is being dialed :) |
| 11:42.31 | jacc0 | asterisk will also be between it when you dial from missed calls |
| 11:43.05 | jacc0 | could you pb cli output of both caals? |
| 11:43.08 | jacc0 | *calls |
| 11:44.09 | jacc0 | there should be absolutle no differends when dialing from missed calls or just dialing the number |
| 11:45.43 | Rolandow | http://pastebin.com/JTjWhLsU |
| 11:46.27 | Rolandow | i don't see the difference :s |
| 11:46.45 | Rolandow | but you must be right, i have canreinvite=no to all sip accounts |
| 11:48.02 | jacc0 | even if it would be set to yes, your sip phone will ask the asterisk server for the ip of the called channel |
| 11:48.32 | Rolandow | ok |
| 11:49.52 | jacc0 | when calling from missed calls it calls this number : Roland_od_Akker-000037ed |
| 11:50.06 | jacc0 | when you dial you dial : 163 |
| 11:50.35 | Rolandow | jacc0: no, that's the originating phone |
| 11:50.38 | Rolandow | it calls Receptie.Fax |
| 11:50.38 | jacc0 | do you have a phone where you can dial chars like 'Roland_od_Akker-000037ed' ? |
| 11:50.52 | jacc0 | if not then don't use alfanumeric sip numbers |
| 11:51.30 | Rolandow | extension number is not the same as sip account |
| 11:51.50 | jacc0 | it is |
| 11:51.58 | jacc0 | at least it could be |
| 11:52.05 | Rolandow | the missed calls log in the phone, stores the sip account, not the extension .. |
| 11:52.08 | Rolandow | in my case it isn't |
| 11:52.15 | kaldemar | it is definitely not the same, nor should it be. |
| 11:52.44 | jacc0 | [100] is dialed by doing `dial(sip/100)` |
| 11:53.01 | Rolandow | jacc0: you have sip accounts and extensions the same .. i don't. |
| 11:53.08 | jacc0 | [alfanumericname] is dialed by doing `dial(sip/alfanumericname)` |
| 11:53.11 | jol | f|shie, Any idea about how to end current AGI-command early, and let it the AGI continue? |
| 11:54.42 | kaldemar | jacc0: exten => 100,1,Dial(SIP/0004f2040001) is the type you should use. |
| 11:55.11 | Rolandow | kaldemar: actually that's what i started it, but because sip accounts show up in display, i decided to use personal sip accounts. |
| 11:55.25 | Rolandow | kaldemar: anyways, that's not the issue :) |
| 11:55.42 | kaldemar | Rolandow: what you should do is define proper caller id names and numbers. |
| 11:55.50 | Rolandow | kaldemar: any idea why i can call by using the direct sip account .. but not trough asterisk |
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| 11:56.32 | Rolandow | kaldemar: yes, i understand why one might use mac addresses .. but i find my logs more readable this way. |
| 11:56.59 | Rolandow | kaldemar: just a matter of taste right? it's not the reason for my problem i think. |
| 11:57.30 | kaldemar | Rolandow: no it's probably not the reason for your issues. the conversation went off the track. |
| 11:57.59 | kaldemar | you should be comparing sip debugs for working and failing calls, not CLI output alone. |
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| 12:04.42 | Rolandow | kaldemar: hmm .. there seems to be something in my subroutine.. when i configure an exten that goes directly to the sip account, it works fine. |
| 12:04.57 | Rolandow | kaldemar: so since it's for fax, i'll just bypass my gosub thingy |
| 12:05.01 | Rolandow | thanks for your time! |
| 12:05.49 | Rolandow | kaldemar: still i'd like to know what a "slic setting" is .. since it seems to be country dependend.. |
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| 12:46.15 | RZero | Hi there, is it possible to use the setting silencethreshold in the dial plan for recording rather than using in |
| 12:46.23 | RZero | it in voicemail.conf |
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| 12:54.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:54.11 | sekil | hello |
| 12:54.24 | sekil | can one use digium card for unframed e1? I guess not.. |
| 12:55.58 | [TK]D-Fender | sekil, Can you clarify "unframed"? |
| 12:56.10 | sekil | full e1..not channelized |
| 12:56.39 | sekil | for example you get E1 link from telco |
| 12:56.42 | [TK]D-Fender | Pretty sure you can.. |
| 12:57.06 | [TK]D-Fender | set it for the usual channel-types in the configs like you do for CAS T1 |
| 12:57.16 | sekil | I'm in europe |
| 12:57.18 | sekil | no t1 |
| 12:57.26 | sekil | I do ccs hdb3 |
| 12:57.31 | [TK]D-Fender | LIKE CAS |
| 12:57.35 | [TK]D-Fender | that isn't the CHANNEL TYPE |
| 12:57.36 | sekil | yeah sorry |
| 12:57.52 | sekil | so I did... |
| 12:58.03 | sekil | I got to the point of working pppd in between two cards... |
| 12:58.12 | sekil | with 30 channels |
| 12:58.15 | [TK]D-Fender | Fell free to show us and describe what you're connecting it to and what that end is set to... |
| 12:58.22 | gusto | sekil: where are you from? |
| 12:58.23 | sekil | and channel 16 doing dig |
| 12:58.26 | [TK]D-Fender | Should get 31 channels |
| 12:58.38 | sekil | but I don't see how to do unframed... |
| 12:59.07 | sekil | right now..this is a channel group actually |
| 12:59.31 | sekil | but I would like to try to get full e1..no channels at all |
| 12:59.49 | [TK]D-Fender | What is the final actual goal? |
| 13:00.17 | sekil | I'm testing a modem line in between |
| 13:00.26 | sekil | to see If I do have 2 meg capability |
| 13:00.48 | sekil | and also to learn these things better :) |
| 13:01.28 | sekil | I see that sangoma has unframed in their wanpipe drivers...not sure how to that with dahdi.. |
| 13:01.44 | [TK]D-Fender | sekil, I'm curious why you would do this at all... who ar you going to hook that up between that there arent much better and cheaper alternatives for? |
| 13:02.19 | sekil | I wouldn't |
| 13:02.25 | sekil | I'm testing |
| 13:02.29 | sekil | and learning like I said.. |
| 13:02.39 | gusto | so you do not have an E1? |
| 13:02.45 | sekil | I do in between |
| 13:02.54 | sekil | two digium cards |
| 13:04.53 | sekil | one is doing clocking for the other |
| 13:07.14 | gusto | sekil: where are you from? norway? |
| 13:07.36 | sekil | does it really matter ? |
| 13:07.43 | gusto | sekil: i am curiois |
| 13:07.45 | gusto | curious |
| 13:07.48 | sekil | ok |
| 13:08.30 | sekil | ok..I guess that unframed is not possible...I see that T1 configs are also channelized |
| 13:08.32 | sekil | thanks |
| 13:08.37 | gusto | looks like serbia |
| 13:09.51 | sekil | ok...now everybody knows.. |
| 13:10.19 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 13:10.23 | gusto | <PROTECTED> |
| 13:10.24 | gusto | <PROTECTED> |
| 13:10.24 | gusto | <PROTECTED> |
| 13:10.26 | gusto | :-D |
| 13:10.34 | gusto | LOL |
| 13:10.46 | sekil | do you mind not doing that? |
| 13:11.06 | gusto | what? |
| 13:11.20 | gusto | dont you know bolek&lolek? |
| 13:11.55 | gusto | i think it is a polish tv show for children |
| 13:11.58 | gusto | cartoon |
| 13:13.21 | sekil | if one sets channels on t1 as clear=1-24 |
| 13:13.50 | sekil | does that imply that the sig channel is 0 |
| 13:13.51 | sekil | ? |
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| 13:13.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:16.28 | [TK]D-Fender | No, there is no straight signalling channel, it uses RBS (robbed-bit) |
| 13:16.53 | [TK]D-Fender | T1 = 24x64bit channels=1544kbps |
| 13:17.01 | [TK]D-Fender | 64kbit* |
| 13:17.17 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 13:17.59 | sekil | it's the signalling inband right |
| 13:18.00 | sekil | ? |
| 13:18.08 | sekil | err inband signalling |
| 13:19.35 | [TK]D-Fender | sekil, Go lookup RBS |
| 13:20.55 | sekil | I did |
| 13:21.23 | sekil | and I concluded there's no separate channel for signalling... |
| 13:21.38 | sekil | I was hoping for a confirmation from north american ppl.. |
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| 13:26.46 | [TK]D-Fender | sekil, where did you read this? |
| 13:29.44 | sekil | http://fengnet.com/book/voip/ch01lev1sec2.html |
| 13:29.57 | sekil | http://modemsite.com/56k/rbs2.asp |
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| 13:32.57 | gusto | vlad_starkov: you want to see something funny? Moscow Bussines Information Center "Binec" <- binec does mean untidiness |
| 13:34.19 | [TK]D-Fender | sekil, So first I tell you how it is supposed to work, then you go look up the results in what apparently has made a printed book, and then ask us AGAIN if this is right? |
| 13:34.35 | [TK]D-Fender | sekil, So you'd like me to say it TWICE? Is 7 good for you? ;) |
| 13:35.14 | sekil | eh? |
| 13:35.22 | sekil | what did I do? |
| 13:36.11 | [TK]D-Fender | "I was hoping for a confirmation from north american ppl.." |
| 13:36.17 | [TK]D-Fender | I told you are the start. |
| 13:36.25 | [TK]D-Fender | Then you found a solid link confiming it. |
| 13:36.33 | [TK]D-Fender | Then you asked again for confirmation. |
| 13:36.34 | WIMPy | The history book of hoorors? |
| 13:36.39 | sekil | (03:16:28 PM) [TK]D-Fender: No, there is no straight signalling channel, it uses RBS (robbed-bit) |
| 13:36.40 | sekil | this? |
| 13:36.43 | sekil | I see it now... |
| 13:36.56 | sekil | sorry really I did not see that.. |
| 13:37.04 | [TK]D-Fender | I'm just wondering how many confirmations you need to feel comfortable for this... |
| 13:37.05 | sekil | my apologies |
| 13:37.40 | sekil | I missed that |
| 13:38.07 | [TK]D-Fender | :p |
| 13:40.14 | sekil | so..I'm getting now 220 kbyte/sec transfer now... |
| 13:40.19 | sekil | hm.. |
| 13:40.40 | sekil | I have 1-5 and 17-31 as clear channels...that should give 1920 kbit/s I guess |
| 13:41.45 | [TK]D-Fender | 1152 <- |
| 13:42.38 | sekil | err 1-15 and 17-31 |
| 13:42.49 | sekil | so 30 channels |
| 13:42.52 | [TK]D-Fender | You shouldn't lose 16 in this scenatio.... |
| 13:42.57 | [TK]D-Fender | scenario* |
| 13:43.19 | [TK]D-Fender | that's normally only euroisdn |
| 13:43.27 | sekil | yep |
| 13:43.37 | sekil | it uses 16 now for sig.. |
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| 13:45.38 | sekil | and I do have it configured for ccs |
| 13:45.53 | sekil | which mandates out-of-band sig |
| 13:46.16 | sekil | hmmm |
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| 14:01.56 | jacobw | ok, i've read up on my issue some more |
| 14:02.16 | jacobw | the leading + in my caller id is uri encoded to %2b |
| 14:02.48 | [TK]D-Fender | jacobw, "pedantic=ys" <- sip.conf |
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| 14:03.08 | jacobw | [TK]D-Fender: what context? |
| 14:03.14 | [TK]D-Fender | [general] <- |
| 14:09.24 | jacobw | i've seen that mentioned on mailing lists |
| 14:09.26 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:09.48 | jacobw | it doesn't work for me, i'm using asterisk 1.6 |
| 14:10.01 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 14:11.23 | [TK]D-Fender | should.... |
| 14:12.14 | [TK]D-Fender | Show us.... |
| 14:13.04 | jacobw | pastebin? |
| 14:13.15 | *** join/#asterisk bipul (3ba2c3e5@gateway/web/freenode/ip.59.162.195.229) |
| 14:13.28 | [TK]D-Fender | yes |
| 14:13.32 | [TK]D-Fender | configs, debug, etc |
| 14:17.57 | jacobw | http://pastebin.com/Jfi5xCkp |
| 14:18.08 | jacobw | ok, there's the SIP messages |
| 14:18.33 | jacobw | i have pedantic=yes in sip_custom.conf included in sip.conf |
| 14:21.31 | [TK]D-Fender | jacobw, Show us actual * CLI SIP debug... |
| 14:21.43 | [TK]D-Fender | jacobw, in full, and the configs... |
| 14:22.31 | jacobw | i can do that, which configs? |
| 14:22.36 | jacobw | sip_*.conf? |
| 14:23.39 | [TK]D-Fender | Everything I need to prove it's i the right place and comms in the right direction... |
| 14:29.29 | bipul | i have Question |
| 14:29.59 | bipul | in Which Fields we can use Asterisk |
| 14:30.44 | drmessano | Strawberry Fields, Forever |
| 14:30.54 | bipul | drmessano: hehehe nice |
| 14:32.26 | jacobw | [TK]D-Fender: debug http://pastebin.com/iP36i43t |
| 14:34.10 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:34.48 | jacobw | [TK]D-Fender: trunk config http://pastebin.com/BjhE5S3Z |
| 14:36.58 | [TK]D-Fender | I don't see the %2b in there .... |
| 14:37.53 | [TK]D-Fender | jacobw, I do see however : Capabilities: us - 0x0 (nothing) <-- that you are offering NO CODECS AT ALL |
| 14:39.15 | [TK]D-Fender | jacobw, And that is an INCOMING call that does not have the encoding you mentioned so there is nothing for it to parse wrong, esp as you aren't even accepting the call and doing anything with it. |
| 14:44.56 | jacobw | [TK]D-Fender: the call is routed to an extension |
| 14:45.12 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
| 14:45.26 | jacobw | [TK]D-Fender: see the contact headers in the 200 messages |
| 14:45.47 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 14:46.08 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 14:46.52 | [TK]D-Fender | jacobw, 488 should kill the call. and I don't see anything proving that the sip debug ebbelongs tot he CLi that follows. And I don't see the %2b in there you say we should be looking for. Also I can tell that you're using freePB and have not shown use REAL configs. |
| 14:49.21 | [TK]D-Fender | jacobw, Where do I see an encoding error in there? |
| 14:49.59 | anonymouz666 | 488 should kill the call -> except if it was a reINVITE ;) |
| 14:52.25 | [TK]D-Fender | anonymouz666, Don't worry, the other 3 layers of masking and BS I'm filtering through make that superfluous :p |
| 14:53.48 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 15:01.06 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 15:02.20 | jacobw | ebbelongs? |
| 15:02.31 | jacobw | ok |
| 15:02.35 | jacobw | i understand |
| 15:03.11 | *** part/#asterisk jacobw (~jacobw@fedora/jacobw) |
| 15:05.35 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 15:12.21 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
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| 15:32.47 | cusco | hello |
| 15:37.05 | cusco | file convert, uses ffmpeg? |
| 15:37.31 | file | nope - it uses the internal media stuff |
| 15:37.42 | file | (internal codecs and that architecture) |
| 15:37.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:37.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:37.53 | cusco | file: does it transcode wma ? |
| 15:37.58 | file | no. |
| 15:38.07 | cusco | ok... |
| 15:42.31 | [TK]D-Fender | cusco, It transcodes things * has format_XXX.so for |
| 15:45.03 | leifmadsen | pabelanger: ohai |
| 15:48.58 | *** join/#asterisk Node_403 (b4ea41ea@gateway/web/freenode/ip.180.234.65.234) |
| 15:49.59 | Qwell | codec_* |
| 15:50.01 | Qwell | file formats don't get transcoded - they get read or written |
| 15:52.06 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 15:52.36 | *** join/#asterisk DerkKo (~afernande@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 15:53.42 | DerkKo | i want to catch "exited non-zero" in the dial plan and do other actions… How can i do it? is it last exten priority plus 101 ? |
| 15:55.12 | *** join/#asterisk oxnard (~oxnard@ws20-ilnk.elkhart.net) |
| 15:56.02 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:56.04 | WIMPy | Wow. Priority jumping was deprecated like five years ago. |
| 15:56.06 | [TK]D-Fender | Qwell, Yeah, codec :) |
| 15:56.27 | [TK]D-Fender | DerkKo, No, "h" standard extension is what you're looking for |
| 15:58.28 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 16:01.31 | DerkKo | LOL. i have not used asterisk in 5 years maybe thats why |
| 16:08.13 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:09.53 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:11.27 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:13.56 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
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| 16:25.22 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:38.08 | [TK]D-Fender | DerkKo, Answer hasn't changed in those 5 years :) |
| 16:38.42 | DerkKo | it has, Priority jumping versus (i,h….) |
| 16:38.44 | DerkKo | :-) |
| 16:39.13 | [TK]D-Fender | Priority jumping was never the answer for "exited non-zero" :p |
| 16:39.32 | leifmadsen | most applications have a ${...STATUS} variable that you can catch and match on |
| 16:39.42 | leifmadsen | ${DIALSTATUS} for example |
| 16:41.39 | DerkKo | i see |
| 16:41.48 | DerkKo | leat me readup on …STATUS |
| 16:42.30 | leifmadsen | note the ... is just a placeholder for the, usually, application name you're using |
| 16:42.33 | leifmadsen | core show application dial |
| 16:42.47 | leifmadsen | that is a good example of useful status variable |
| 16:45.42 | DerkKo | cool |
| 16:45.54 | DerkKo | thanks for pointing me in the right direction. |
| 16:49.39 | *** join/#asterisk _pll (~melisa@190.235.77.251) |
| 16:50.01 | _pll | Hello, is there a workaround for this issue? https://issues.asterisk.org/jira/browse/ASTERISK-18945 |
| 16:50.37 | WIMPy | If it's not listed there, obviousely not. |
| 16:50.45 | _pll | Can I set ignorebusy=no on the sip user and will it be called even if busy? |
| 16:52.23 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 16:54.00 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 16:55.55 | *** join/#asterisk wkemp (~wkemp@adsl-072-156-217-048.sip.bna.bellsouth.net) |
| 16:56.34 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 16:57.26 | sekil | [TK]D-Fender: I managed to pass 30x64 kbits on e1/ccs...1920kbit/s |
| 16:57.34 | sekil | [TK]D-Fender: thanks for your help |
| 16:58.33 | [TK]D-Fender | sekil, You're welcome |
| 16:59.09 | sekil | [TK]D-Fender: as I said before...I couldn't do unframed e1..but nm |
| 16:59.54 | WIMPy | I'm not sure if I shoud dare to ask, but what are you doing there? |
| 17:05.42 | wkemp | I'm a babe in the woods here, but my pbx vendor and data / voice vendor are leaving me in a lurch... so i wanted to throw this out here and see if anyone has run across an issue before. Running a managed pbx with Asterisk 1.6.2.21 under a vendor's webgui. Single PRI talking to a AT&T managed Cisco router. Outbound calling shows absolutely no issues. Intermitently, inbound calling will drop with callers getting a |
| 17:05.42 | wkemp | <PROTECTED> |
| 17:06.17 | sekil | I was trying to do full e1 data link across one pair of dsl modems with digium cards at both ends |
| 17:07.09 | WIMPy | sekil: Ah. And you couldn't use the full interface? |
| 17:08.55 | WIMPy | wkemp: Do you have a high call volume? That sounds like the "glare" issues with libpri. |
| 17:09.47 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 17:09.59 | sekil | WIMPy: no..only channelized |
| 17:10.13 | wkemp | WIMPy: not what i'd call high. We will average about 6 active calls during most of the day. Single pri, outbound start 23 - down , inbound 1-up 24 for signaling |
| 17:10.42 | WIMPy | wkemp: Sounds unlikely then. |
| 17:11.04 | WIMPy | sekil: I guess that's due to the modems then? |
| 17:11.09 | wkemp | during the last 2 hour finger pointing session glare was talked about briefly and dismissed because of the improbability of overlap |
| 17:11.20 | sekil | WIMPy: no |
| 17:11.33 | WIMPy | oh |
| 17:11.50 | sekil | WIMPy: I don't see a way to do unframed e1 with dahdi/digium...modems are ok.. |
| 17:12.45 | WIMPy | As you have to configure all sorts of stuff, you usually wouldn't care about, I would have assumed that to be possible, |
| 17:18.49 | [TK]D-Fender | wkemp, inbound gets "all circuits busy"? That measn you're describing debug from an *-based CALLING system. You should show us the actual complate call attempts for this.... |
| 17:18.53 | [TK]D-Fender | ~pb |
| 17:18.53 | 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. |
| 17:18.55 | [TK]D-Fender | ^^^ |
| 17:19.17 | [TK]D-Fender | wkemp, and that message by itself is not meaningful. Further debug is required |
| 17:19.54 | sekil | WIMPy: I don't think so...you have to specify some framing for span |
| 17:22.54 | wkemp | [TK]D-Fender : here's at&t's dump to me. I don't have anything from the asterisk side at the moment |
| 17:22.56 | wkemp | http://pastebin.com/Fzz8rYjb |
| 17:23.44 | [TK]D-Fender | wkemp, PB : "dahdi show status", "dahdi show channels", etc |
| 17:23.54 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 17:24.32 | wkemp | [TK]D-Fender, its a managed pbx so I don't have access to get the data. I was just posting as much info as i could in hopes i might find someone who's seen the issue before. |
| 17:24.34 | WIMPy | That doesn't say all circuits busy. It says requested channel not available. |
| 17:24.46 | [TK]D-Fender | ^^ |
| 17:24.59 | [TK]D-Fender | wkemp, And if you're running blind.. well there's even less we can do |
| 17:25.38 | wkemp | fair enough, i thought it was worth a shot before i fire the pbx vendor and tear down the box and build the whole thing back up again. |
| 17:25.44 | *** join/#asterisk ruunix (~ruunix@190.196.71.206) |
| 17:25.48 | WIMPy | And some 3rd party config tool makes the whole thing nearly impossible. |
| 17:25.51 | ruunix | hi there. . |
| 17:26.21 | WIMPy | If it's managed by someone else, why do you care? |
| 17:26.40 | WIMPy | Get them to fix it. |
| 17:26.47 | ruunix | little question .. softphone without g729, but provider needs g729 .. what can i do .. is there a possible to register a sip inside the server , to convert g729 to another codec ? |
| 17:26.56 | wkemp | they'v ebeen trying since april |
| 17:27.22 | wkemp | its degenerated into at&t saying "blame the pbx vendor" and the pbx vendor saying "blame at&t" |
| 17:27.23 | WIMPy | Then get a lawyer and become rich :-) |
| 17:27.40 | WIMPy | ruunix: You can by G.729 liceses from Digium. |
| 17:28.21 | WIMPy | Well, from that debug it's clear that your end doesn't accept the calls. |
| 17:28.52 | wkemp | right... "my" end thinks the channel is in use.. its like it isn't getting the message that teardown of the call occured. |
| 17:29.32 | wkemp | maddening thing is this happens once or twice every two weeks |
| 17:29.35 | WIMPy | So unless that Cisco in the middle is doing somethign strange... Is that owned/managed by AT&T? |
| 17:29.48 | wkemp | the cisco is owned and managed by at&t yes |
| 17:30.03 | wkemp | their first crack at the problem was to replace a failing fan assembly |
| 17:30.09 | wkemp | *hangs head* |
| 17:30.26 | WIMPy | You need to get debug from te Cisco or the Asterisk Box to be sure. |
| 17:31.18 | ruunix | how can i reboot asterisk without reboot calls ? |
| 17:31.25 | WIMPy | But seriousely: The Asterisk Version seems too old to bother with anyway. Don't use anything befor 1.8 with ISDN. |
| 17:31.40 | WIMPy | What? reboot without reboot? |
| 17:31.44 | wkemp | yeah, i was a little surprised they were back on 1.6 |
| 17:31.51 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 17:32.15 | wkemp | i'm sure it has to do with their little webgui breaking on 1.8 in some form or fashion |
| 17:32.40 | ruunix | mmmmm ... i make changes in sip.conf and queues.conf and extensions.conf.. but i dont wanna hangup all the calls ... |
| 17:32.41 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 17:32.49 | WIMPy | Quite possible, but that's SEP. |
| 17:32.55 | ruunix | to take the new configuration .. |
| 17:33.09 | ruunix | is there a way to restart just the configs ? |
| 17:33.17 | ruunix | without hangup the calls ? |
| 17:33.36 | WIMPy | 'sip reload'. Don't know about queues. Probaly the same. |
| 17:34.01 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 17:35.13 | ruunix | thanks .. just extensions are different |
| 17:35.36 | WIMPy | 'dialplan reload' for extensions. |
| 17:37.15 | ruunix | thank you very much |
| 17:38.11 | wkemp | on a totally different note, how common is it to run asterisk in a ESXi hosted vm and do a hardware passthrough for interface card ? |
| 17:42.51 | [TK]D-Fender | wkemp, Asking for trouble |
| 17:43.24 | wkemp | because of the hardware pass-through ? |
| 17:43.36 | [TK]D-Fender | yes |
| 17:44.43 | wkemp | [TK]D=Fender, what are some of the issues that arise? Timing problems? |
| 17:45.57 | *** join/#asterisk bjweeks (~bjweeks@wikipedia/Brandon) |
| 17:52.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:52.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:56.42 | *** join/#asterisk _val_ (fbsd@178-84-212-245.dynamic.upc.nl) |
| 17:59.50 | *** join/#asterisk Node_403 (b4ea4f13@gateway/web/freenode/ip.180.234.79.19) |
| 17:59.53 | gusto | is IAX2 trunking automatic, or do i have to configure it somewhere? |
| 18:02.34 | sawgood | automatic IAX2 trunking, where is that script? |
| 18:07.15 | sekil | there was trunk=yes param back in 1.2 |
| 18:08.48 | JamKo | Anyone shed some light on these errors during MAKE of asterisk 1.8.13.0? : http://pastebin.com/8jWuRD5X |
| 18:09.56 | _val_ | hey eveyrone. I've these lines in the config file. http://pastie.org/4109846. If the callerID is larger (has more number) I need to enter more XXX'es. How can I change this so it automatically detects CallersID? |
| 18:12.42 | sekil | _val_: you mean this dialplan doesn't match the called party? |
| 18:12.58 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-cksmoypslhzwgwzm) |
| 18:13.11 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-rnrxgnefbykjmdya) |
| 18:13.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:13.53 | _val_ | sekil: yes. So if the call I'm going to make has 12 digits.. I need to enter 12 XXXX...'es... to make it work |
| 18:14.10 | Hive | _0X. |
| 18:14.21 | sekil | _val_: so what's the connection than with callerid? |
| 18:14.30 | sekil | _val_: just use 0X. |
| 18:14.45 | sekil | it will match 0whatever |
| 18:14.52 | Hive | need that _ at the beginning, no? |
| 18:15.08 | sekil | that is ...0[digit]whatever |
| 18:15.24 | _val_ | sekil: http://pastie.org/4109872 Like this? |
| 18:15.43 | _val_ | exten=_0X.,3.... |
| 18:15.55 | sekil | yeah..that would match all after 0[digit] |
| 18:15.58 | _val_ | is it correct. Notice (.,) after X.. |
| 18:16.06 | sekil | yes it is correct |
| 18:16.12 | _val_ | Allright. Thanks a ton guys! |
| 18:16.12 | leifmadsen | . means 1 or more digits |
| 18:16.18 | leifmadsen | ! means 0 or more digits |
| 18:16.23 | gusto | yes |
| 18:16.32 | gusto | and what was about that trunking? |
| 18:16.38 | leifmadsen | _val_: I suggest reading the Dialplan Basics chapter at asteriskdocs.org |
| 18:16.47 | _val_ | leifmadsen: thank you too. Makes more sense. |
| 18:16.50 | _val_ | going to do so. |
| 18:17.20 | sekil | gusto: trunk=yes in iax.conf |
| 18:19.50 | gusto | hm |
| 18:19.55 | gusto | sekil: have you tried it? |
| 18:20.13 | sekil | gusto: using it for years |
| 18:20.15 | gusto | sekil: does it work like i imagine, that it uses only one connection and adds all calls into it? |
| 18:20.35 | WIMPy | It doesn't use connections at all. |
| 18:20.52 | WIMPy | And it always only uses one port, no matter if you enable trunking or not, |
| 18:20.53 | gusto | sekil: well, i am relatively new to asterisk, started this year and i am going to try IAX2 |
| 18:21.01 | gusto | ha? |
| 18:21.04 | gusto | how's that? |
| 18:21.25 | WIMPy | That's the way it was designed. |
| 18:21.27 | sekil | gusto: it makes a trunk...adds 4 byte header..and putting all payloads inside one big chunk |
| 18:22.01 | gusto | i understanded trunking so that it omits beaurocracy around protocols by putting the audio streams into that one IAX connection instead of a lot of RTP streams |
| 18:22.23 | WIMPy | IAX doesn;t use RTP. |
| 18:22.49 | WIMPy | And yes, it's about only sending those headers once for all channels. |
| 18:23.06 | sekil | gusto: there's no signalling per call |
| 18:23.53 | sekil | gusto: it multiplexes all the calls in one ....with using compressed codecs bw needed is amazingly small |
| 18:23.55 | WIMPy | Don't mix terms. Signalling is always per call. |
| 18:24.17 | *** join/#asterisk Node_403 (b4ea4f13@gateway/web/freenode/ip.180.234.79.19) |
| 18:24.40 | WIMPy | And the used CODECs don't care if they are transported via IAX or RTP. |
| 18:25.33 | sekil | WIMPy: I find iax trunking with g729 not the same in bw as sip+rtp+g729 |
| 18:25.49 | WIMPy | That's because of the trunking in IAX. |
| 18:26.09 | *** part/#asterisk _val_ (fbsd@178-84-212-245.dynamic.upc.nl) |
| 18:26.16 | sekil | yeah...which is why I said the bw usage is terribly small |
| 18:26.21 | WIMPy | Only one set of (minimally bigger) headers, instead of headers for each channel. |
| 18:26.37 | sekil | I have coleagues pushed 200+channels on little alone 1meg line |
| 18:26.52 | WIMPy | And when using low bitrate CODECs the difference can become massive. |
| 18:26.53 | sekil | or something in between 1-2 meg... |
| 18:27.22 | sekil | 8kbit/s per call..no ethernet..ip..udp..header... |
| 18:27.24 | sekil | just great |
| 18:29.12 | DerkKo | is there a meet me UI that allows users to manipulate active conferences? Such as active callers, Mute unmute. Kick and dial in users ? |
| 18:31.59 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-cswvwyjhaqzhamrm) |
| 18:33.39 | *** join/#asterisk bjweeks (~bjweeks@wikipedia/Brandon) |
| 18:45.15 | gusto | yes, and also he said that it multiplexes all audio streams, what happens in meaning of a DVB-T MUX and NOT asterisk MUX, because that would be a conference call |
| 18:51.21 | sekil | gusto: ? |
| 18:51.57 | *** join/#asterisk cmnky (~cmnky@unaffiliated/cmnky) |
| 18:52.47 | gusto | sekil: dont you have DVB-T in serbia? |
| 18:53.42 | cmnky | I recently switched over to verizon fios digital voice, and i'm getting tons of crap/spam/scam/telemarketer phone calls ... can someone advise on an inexpensive method to use asterisk to block these calls, but let non-blacklisted calls pass through ? i can read and deal with the software alright, but not sure about the hardware / card required to do this for 2 phone lines ( they act like POTS lines comming out of the ONT ) |
| 18:56.28 | *** join/#asterisk Michaelb8225 (46441f88@gateway/web/freenode/ip.70.68.31.136) |
| 18:58.46 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:58.51 | Michaelb8225 | in voicemail.conf you specify the extension, then the password, an email address (from Lief's book) like so: 100 => 1234,Leif Madsen,leif@shifteight.org ... and then in the extensions.conf you setup VoiceMailMain() like this: exten => 8500,1,VoiceMailMain(). Given all that, how come when I call 8500, enter extension 100, and password 1234, it says the password is incorrect? |
| 18:59.54 | bjweeks | cmnky: easiest way is going to be a ATA, a FXO/FXS card is more complicated/expensive. |
| 19:00.20 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 19:00.59 | cmnky | bjweeks, i have a linksys single line ATA box ... but how is that going to answer a POTS line and ring through if not blacklisted ? |
| 19:01.37 | cmnky | its an SPA2102 .. if that means anything to you |
| 19:01.44 | bjweeks | cmnky: pass the call to an Asterisk server, then use Asterisk to manage the blacklist |
| 19:01.48 | cmnky | which is actually a 2 line box i guess |
| 19:02.51 | cmnky | so this box will answer the call, then validate against an asterisk server over ethernet ... then what ? how does it ring the POTS phones on the line it just answered ... can it do that ? or do i need to change the house over to VoIP phones ? |
| 19:04.07 | cmnky | or do i tell the box to answer *if* the call is blocked, otherwise, don't answer and let the POTS phones ring ? |
| 19:06.00 | cmnky | or can these boxes be used to ring through from line 1 to line 2 if someone enters a keycode ... and i would need 2 of these boxes ... one for each phone line ? |
| 19:06.21 | bjweeks | cmnky: Ah, I understand. If you had a three port ATA (or more than one) you'd take in the two phone lines into two the ATA and then use port to connect to your home wiring |
| 19:06.49 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-9.phlapa.fios.verizon.net) |
| 19:07.35 | cmnky | well, this box has phone 1, phone 2, ethernet, and internet ports ... could i connect phone 1 to the ONT ( telco line ) and phone 2 to the house wiring, and then selectively ring through ? |
| 19:07.50 | bjweeks | Yup! |
| 19:07.53 | cmnky | sorry if these are n00b questions ... im a programmer, and I know about voip .. just have 0 experience with it |
| 19:08.36 | cmnky | bjweeks, alright ... super ... i'll put the project on my to-do list then ... and yeah, since i have the box, way cheaper than a PCIe card ... so i'll leave that for another time after i have asterisk experience ... ty ty |
| 19:10.32 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 19:10.36 | cmnky | thanks again |
| 19:10.38 | *** part/#asterisk cmnky (~cmnky@unaffiliated/cmnky) |
| 19:12.04 | Michaelb8225 | anyone? |
| 19:13.05 | bjweeks | Michaelb8225: log output? |
| 19:13.35 | Michaelb8225 | the CLI just says incorrect password |
| 19:14.43 | bjweeks | did you reload app_voicemail? |
| 19:18.59 | *** join/#asterisk Michaelb8225 (46441f88@gateway/web/freenode/ip.70.68.31.136) |
| 19:19.59 | Michaelb8225 | bjweeks -- sorry I guess the webclient errored out... Yes, I did a core reload. I think that reloads voicemail, no? I'm new to *... trying to run off the 'definitive guide' book |
| 19:21.27 | bjweeks | Michaelb8225: Not sure to be honest, I always use 'module reload app_voicemail.so' after VM changes, mass reloads on production systems are a bad idea. |
| 19:21.53 | Michaelb8225 | no doubt. This is still very much a test box, so... I'll try that and see what happens |
| 19:23.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:23.32 | Michaelb8225 | nope. tried two different mailboxes, with different passwords... asks for mailbox, says pw is incorrect, lets me try 3 times and hangs up |
| 19:23.56 | Michaelb8225 | one additional detail (if it makes any difference) I'm using a Zoiper softphone for testing |
| 19:25.12 | bjweeks | Do you see the line wrong password 'xxxx' for extensions 'xxxx' in the logs? Make sure both those numbers match up. |
| 19:26.07 | Michaelb8225 | -- Incorrect password '4096' for user '100' (context = default) |
| 19:27.01 | Michaelb8225 | which logs are you referring to? |
| 19:28.53 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 19:29.46 | bjweeks | Michaelb8225: Is your VM context default? Only think I can think of. |
| 19:32.23 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 19:32.42 | edge | How do i reload the IAX configuration ? |
| 19:33.26 | bjweeks | 'iax2 reload'? |
| 19:34.10 | edge | well then my questions grows larger |
| 19:34.22 | edge | iax2 doesn't seem to be a valid command at the cli |
| 19:34.55 | edge | its worth noting i just wrote the iax.conf while asterisk was running |
| 19:35.43 | bjweeks | 'module load chan_iax2.so' |
| 19:35.51 | edge | bjweeks: perfect. Thanks |
| 19:41.41 | Michaelb8225 | that might be it. |
| 19:41.51 | Michaelb8225 | how do you define the context? |
| 19:43.01 | bjweeks | It'll be the name in [brackets] above your definition lines in voicemail.conf |
| 19:43.57 | Michaelb8225 | right so that is defined there... how do I define it in the dialplan? |
| 19:45.09 | *** join/#asterisk digits (~digits@213.219.154.252.adsl.dyn.edpnet.net) |
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| 19:47.41 | digits | Hi all, I have some trouble getting asterisk to work , it's described in detail here: http://stackoverflow.com/questions/10951472/asterisk-nat-either-get-called-or-be-called . Is there anybody who think they could help me ? I'm willing to compensate you for your trouble. |
| 19:49.07 | Qwell | 1000-2000? |
| 19:49.24 | Qwell | Perhaps you mean 10000-19999? |
| 19:49.44 | digits | yes, that's what i meant |
| 19:50.05 | *** join/#asterisk techyjt (~techyjt@adsl-74-176-124-127.asm.bellsouth.net) |
| 19:50.12 | digits | either way, i can make phone calls, and i can also receive them, depending on the directmedia yes/no option, but not at the same time |
| 19:50.26 | digits | and that's something that i'm trying to fix |
| 19:51.54 | digits | TCP / UDP 10000-20000 and TCP/UDP 5060 and 4569 are forwarded to my asterisk server |
| 19:53.40 | Qwell | You need to set localnet. You also need to upgrade. |
| 19:53.43 | Qwell | ~upgradeasterisk |
| 19:53.43 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:55.59 | techyjt | Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk samples |
| 19:55.59 | techyjt | so I realize that may have messed me up. However, I need a little help just getting the basic dahdi and dialplan to simple get the analog phone to work. Can someone spare a little time? |
| 19:56.15 | digits | I can't upgrade asterisk. The software is running on a modified NAS server, which probably doesn't support a more recent asterisk version |
| 20:00.48 | techyjt | Anyone? I am googling a lot but every example I have found sets up SIP softphones etc. I can't even get dialtone on the analog phone. |
| 20:01.50 | jaytee | ~book |
| 20:01.50 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:02.48 | jaytee | wow, I never realized Linus had such a potty mouth! :-) |
| 20:04.18 | leifmadsen | eh? |
| 20:04.36 | Get_The_Fish | does anyone know if there is going to be an official Centos 6 yum repo for asterisk, or where I can find a good unofficial one? |
| 20:06.35 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:12.05 | Get_The_Fish | is there any way to force yum to use the centos5 repos instead of centos6? |
| 20:13.54 | Get_The_Fish | any community leaders know if the org will give me server space if I want to maintain it? |
| 20:15.11 | jaytee | leif, Linus gave a speech in Finland last week and at the end of it he said Nvidia was the worst company ever and then said "F&%K you, NVIDIA" and flipped the middle digit. |
| 20:15.33 | Get_The_Fish | and that would be because he is right jaytee |
| 20:16.20 | Get_The_Fish | you know how many years those assholes have comsumed while people reverse engineer their adapters so we can get some driver support that doesn't suck? |
| 20:16.31 | jaytee | Get_The_Fish, yeah, I spent several hours getting an nvidia card to work at 1440x900 on Ubuntu 6.06 years back. Took the same dual boot machine 20 seconds and two mouse clicks on Windows Vista. |
| 20:17.03 | Get_The_Fish | jaytee, you and lots and lots of others. Imagine the poor hacker that has to get those bug reports. |
| 20:17.07 | jaytee | 2112, The Year of the Linux Desktop! |
| 20:17.15 | Get_The_Fish | ROFL |
| 20:18.21 | jaytee | after playing with Ubiquity and Gnome 3 I wanna go backwards not forwards |
| 20:18.40 | Get_The_Fish | you and lots of others |
| 20:19.25 | Get_The_Fish | I was on the list for GNOME 3, and I can tell you from experience that bad community leadership yields bad products, and GNOME 3 is a perfect example. |
| 20:20.18 | Get_The_Fish | You could watch the developers just ignore the community and stick to this very dogmatic approach to the design, which alienated everyone. By the textbook the UI is perfect, thats why it's so terrible. |
| 20:20.47 | leifmadsen | actually likes Gnome 3 |
| 20:21.08 | Get_The_Fish | then you would have really liked the first few iterations. |
| 20:22.11 | Get_The_Fish | I used it in almost alpha mode for over a year, and over a year on the desktop after the release, and it has gotten more obtuse not less IMO. |
| 20:26.28 | Get_The_Fish | so leif whats up with the yum repos for centos 6? If no one else will maintain it I will... I just need the resources |
| 20:28.41 | jaytee | all I wanted from Gnome 3 was a sound event for closing a window so I could use a clip of Talking Heads "Burning Down the House" with the fire effect in Compiz Fusion. |
| 20:29.19 | [TK]D-Fender | jaytee, There is water ... at the bottom of the ocean! |
| 20:29.29 | jaytee | lol |
| 20:29.47 | jaytee | that is not my beautiful wife! |
| 20:30.24 | [TK]D-Fender | I was close to adding that to my performance roster... I picked up "And She Was" on keys for my latest band, but that direction is a little fixxled |
| 20:30.41 | leifmadsen | Get_The_Fish: sorry, I'm not in charge of any of the yum/rpm stuff -- there are rpms already being created and maintained via fedora. Might be able to use those resources. Otherwise check in with Qwell |
| 20:31.50 | Get_The_Fish | excellent, thanks leif. The thing about those fedora builds is that the deviation on some of those packages from the Centos base is getting quite large, and I'm concerned about bin compat. |
| 20:32.08 | Get_The_Fish | I will check with Qwell though, thanks. |
| 20:32.21 | [TK]D-Fender | ok, checkout time over here, BBIAB |
| 20:33.05 | leifmadsen | Get_The_Fish: yep I hear ya, just not much info I can offer |
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| 21:02.06 | Get_The_Fish | hmmmm, epel huh |
| 21:03.10 | techyjt | anyone got some time to help with asterisk tdm410 analog setup? Any web pages or anything? I am simply trying to get a POTS line and a analog phone working. |
| 21:03.36 | Get_The_Fish | whats wrong with it? |
| 21:06.52 | techyjt | I can't even get dial tone through it. I have tried several example dialplans and just not able to really find any information to get a simple pbx setup. Most everything I find wants SIP softphones, ext. I compiled it with asterisk samples so maybe some of those files have messed me up? |
| 21:12.35 | Get_The_Fish | I'm assuming that the line is a FXS line. When you say dial tone, are you saying dial tone on the softphones or the pots line? |
| 21:14.17 | techyjt | Yes, I have 2 FXO with 1 POTS line in and 2 FSX with 1 analog phone connected. I mean dialtone on the analog phone. I pick up the phone connected to port (3) the POTS line is in port (1). I guess it is something simple but I am just starting out. |
| 21:15.06 | techyjt | I meant it gets not dial tone. |
| 21:15.51 | [TK]D-Fender | then either you didn't configure the port right, didn't initialize DAHDI, or you forgot to plug in the molex to the card |
| 21:15.57 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:16.02 | Get_The_Fish | so dialtone is generated off the card in that case I believe, so yeah what TK said |
| 21:17.07 | techyjt | The card is powered and has all the lights on. So I should re-initalize DAHDI? |
| 21:17.48 | Get_The_Fish | I would check that conf file over a few times first, make sure that you have all your eyes dotted and your tees crossed |
| 21:18.45 | techyjt | Okay, I will try again. Thanks. |
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| 22:19.58 | ThinkGNU- | Can anyone tell me the best way to send text messages to SIP phones |
| 22:19.59 | ThinkGNU- | ? |
| 22:22.09 | ThinkGNU- | I tried SendText() but that doesn't seem to be doing it |
| 22:27.52 | [TK]D-Fender | T.38 isn't a codec....sipsak <- |
| 22:27.56 | [TK]D-Fender | * is not meant for this |
| 22:28.03 | [TK]D-Fender | oops... |
| 22:28.09 | [TK]D-Fender | sipsak <- |
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| 22:46.29 | *** join/#asterisk MichaelB8225 (46441f88@gateway/web/freenode/ip.70.68.31.136) |
| 22:46.41 | ThinkGNU- | Are there only certain phones that can receive text messages on their LCD screen? |
| 22:47.22 | MichaelB8225 | quick question, are there any good Asterisk system sizing guides? I were planning a system of say... 250 people with 40 concurrent calls... how would I plan for how many asterisk servers and so on? |
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| 22:54.50 | ACiDV | Hi all! I'm using MixMonitor and Set(PITCH_SHIFT=xx), does it normal that the monitor file record the unchanged voice ? |
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| 23:04.54 | *** join/#asterisk sam555 (~sysadmin@udp124488uds.hawaiiantel.net) |
| 23:05.01 | sam555 | hello all! |
| 23:05.11 | sam555 | what version of freepbx does asterisk now 2.0.2 32 bit have? |
| 23:08.57 | [TK]D-Fender | MichaelB8225: Any modem PC should handle that fine |
| 23:09.01 | [TK]D-Fender | modern* |
| 23:09.13 | MichaelB8225 | one box? |
| 23:09.18 | [TK]D-Fender | Yes |
| 23:09.45 | [TK]D-Fender | Remember that Digium was selling quad E1 cards over a decade ago. that's 120 channels <- |
| 23:09.50 | MichaelB8225 | is it adviseable to move other services to other machines, things like MeetMe() and voicemail? |
| 23:10.03 | [TK]D-Fender | That piece of the math should be pretty obvious |
| 23:10.16 | [TK]D-Fender | Vm isn't a real load. |
| 23:10.24 | WIMPy | MichaelB8225: If you plan to use wrist watches as servers, yes. |
| 23:10.30 | [TK]D-Fender | Meetme... is a question depending on exact circumstances |
| 23:11.29 | WIMPy | I'd go for ConfBridge instead of MeetMe. |
| 23:13.27 | MichaelB8225 | Sorry, its not obvious. I've been doing some searching... and even this book here suggests that "large" systems - over 120 phones - should be spread across multiple systems... and thats only a year old. Other sources indicate it should be fine, other sources indicate it should be fine as long as we were to offload things like conferencing. |
| 23:13.49 | [TK]D-Fender | Which book? |
| 23:14.15 | MichaelB8225 | Digium's site suggests Google and Ebay use Asterisk (cool), but don't provide any manner of information on how many systems or how many users, calls etc. |
| 23:14.33 | MichaelB8225 | Asterisk: The Definitive Guide, 3rd edition |
| 23:14.51 | [TK]D-Fender | what chapter? |
| 23:15.27 | WIMPy | The number of concurrent calls is not really relevant. It's what you do to them. |
| 23:15.41 | WIMPy | DTMF detection, recording, transcoding. |
| 23:16.00 | jpsharp | Its recording and transcoding that eats system resources. |
| 23:16.06 | MichaelB8225 | Appendix C, pg's 645 & 646 |
| 23:16.10 | WIMPy | That's why you don't find any numbers anywhere. |
| 23:18.10 | [TK]D-Fender | No mention of PCI-X, etc |
| 23:18.37 | [TK]D-Fender | MichaelB8225: Remember ... the Appendix is an organ whose sole purpose is to KILL you when it fails :) |
| 23:19.02 | [TK]D-Fender | MichaelB8225: Looks like they just carried that chapter over wholesale from years ago |
| 23:19.22 | [TK]D-Fender | dual-Xeon locomotive |
| 23:19.27 | [TK]D-Fender | Whee! |
| 23:19.37 | [TK]D-Fender | Cause yeah, that's like an i7 today! |
| 23:19.42 | [TK]D-Fender | only... not at all :p |
| 23:20.07 | [TK]D-Fender | trancoding is a load... recording... shouldn't be if you have a brain :) |
| 23:20.26 | MichaelB8225 | fair enough. Thing is I'm trying to size up a phone system for a 250 person financial institution spread across a half dozen or more locations, lots of remote workers (like me, I work from home)... and at some point I'm going to have to tell Management how much this is going to cost... am I going to need a couple of small servers or a bunch of beefy ones. |
| 23:20.56 | MichaelB8225 | if I go tell the Boss I need 6x $8000 servers, plus a bunch of PRI cards... it might be cheaper to knock on Mitel's door |
| 23:21.12 | [TK]D-Fender | Going to record a lot? What codecs involved? |
| 23:21.21 | MichaelB8225 | record? prob not. |
| 23:21.25 | MichaelB8225 | codecs? simple ones. |
| 23:21.34 | [TK]D-Fender | That sounds like a sing 300$ PC + card to me. |
| 23:21.36 | [TK]D-Fender | TOPS |
| 23:21.40 | [TK]D-Fender | $3000 |
| 23:21.48 | [TK]D-Fender | Even that is generous |
| 23:22.00 | MichaelB8225 | none of that has been nailed down yet. Right now its in the very testing stage. I've got a VMWare VM running Ubuntu and some softphones |
| 23:22.06 | jpsharp | And that's only 3K so you can get a machine with RAID + dual power supplies |
| 23:22.23 | MichaelB8225 | trying to convince them to get me a few Polycom hardphones for testing. |
| 23:23.22 | MichaelB8225 | They might want to setup an IVR... for personal lending press 1, for commercial lending press 2, for all other inquiries, please go to hell |
| 23:23.22 | [TK]D-Fender | eek |
| 23:23.40 | jpsharp | bgjuhou8jpijk0-=8iuj/ |
| 23:23.46 | jpsharp | Damn cat |
| 23:23.47 | MichaelB8225 | eek? |
| 23:23.59 | [TK]D-Fender | yeah, my PC just tried to sleep on me |
| 23:24.05 | [TK]D-Fender | Didnt hink IRC would survive |
| 23:25.30 | carrar | espresso |
| 23:25.37 | carrar | pour it all over the keyboard |
| 23:25.48 | MichaelB8225 | I know Digium makes phones too (and I'm not against digium hardware) but our vendor for "stuff" carries Polycom, they have free shipping and don't need to cross the border from the US |
| 23:25.57 | MichaelB8225 | and Polycom makes good stuff. |
| 23:26.52 | jpsharp | I haven't tried the Digium phones, but I've used Polycom IP300/301s and Cisco 7940s with excellent results. |
| 23:27.08 | carrar | We've tested a little on the Digium phones |
| 23:27.15 | carrar | they still need some work |
| 23:27.29 | MichaelB8225 | ok cool, Polycom it is. last two things to replace Skype (yay!): 1. when we tried doing a call from softphone to softphone, both of us had laptops and we were using the built in mic and speakers. We got some nasty feedback. Skype lets you use your built in stuff... what am I doing wrong? 2. Video conferencing? |
| 23:33.00 | gusto | MichaelB8225: what softphone are you using? |
| 23:33.10 | MichaelB8225 | Zoiper communicator |
| 23:33.26 | gusto | MichaelB8225: have you enabled video in asterisk? what protocol do you want to use for video transport? |
| 23:33.41 | gusto | MichaelB8225: is it a windows application? i do not know that! |
| 23:33.42 | MichaelB8225 | just the free one for now. I'm not tied to it... but we do have a Mac user, and it supports iOS so... |
| 23:34.13 | gusto | mac user? iOS? where? on a iPad or something? |
| 23:34.19 | MichaelB8225 | Protocol? I don't really care... whicheverone works. |
| 23:34.24 | MichaelB8225 | sorry, not iOS... OS X |
| 23:34.28 | gusto | ah |
| 23:34.38 | gusto | well, on macosx there are a lot of apps working |
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| 23:36.32 | gusto | does not look bad and it supports also IAX protocol ... hopefully it's the IAX2 and also SIP2 :-D |
| 23:36.51 | ACiDV | sip2 ? |
| 23:36.58 | MichaelB8225 | yeah, it uses IAX2. thats how I got around the problems inherent with NAT + SIP |
| 23:37.27 | gusto | MichaelB8225: have you tried iChat? that supports SIP and would work best |
| 23:37.52 | MichaelB8225 | iChat would work... but SIP through NAT sucks. |
| 23:37.54 | gusto | ACiDV: SIP v2 |
| 23:38.02 | WIMPy | SIP v2857 |
| 23:38.25 | ACiDV | ok ;P |
| 23:38.28 | gusto | MichaelB8225: can't you just use port-forwarding? |
| 23:38.31 | WIMPy | Maybe SIP v14735 will be usable. |
| 23:40.06 | gusto | This version of the protocol was further refined to version 2.0 and clarified in RFC 3261, although some implementations are still relying on the older definitions.[specify] |
| 23:40.10 | WIMPy | And together wis Asterisk 23 we might have a goot PBX replacement. |
| 23:40.14 | MichaelB8225 | Sure I could use port forwarding. Now I have to explain to 200 users, each with a different consumer router device at home that wants to us a softphone how to either port forward or enable SIP pass-through. Its great, but not so awesome. |
| 23:40.22 | WIMPy | Did you notice I'm in a good mood today? :-) |
| 23:40.22 | MichaelB8225 | IAX2 just... worked. |
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| 23:41.17 | WIMPy | likes IAX, but it could need a little more TLC today. |
| 23:42.02 | gusto | MichaelB8225: yes, i am having the same problem, ppl just do not touch their routers because they fear to "break" something |
| 23:42.12 | gusto | maybe H323 would be a solution to that |
| 23:42.50 | WIMPy | H.323 uses RTP, just like SIP. |
| 23:43.12 | WIMPy | But otherwise it is excessively better. |
| 23:44.22 | gusto | heh? h323 uses RTP? |
| 23:45.31 | MichaelB8225 | in a previous environment, I had Micro$oft Lync running. That could do phone calls, pc-to-pc audio or A/V conferences... but... costs $$, and I'm told not such a great device for telephones. |
| 23:45.33 | gusto | but it seems that you are right |
| 23:45.41 | gusto | H.323 was the first VoIP standard to adopt the Internet Engineering Task Force (IETF) standard Real-time Transport Protocol (RTP) to transport audio and video over IP networks.[citation needed] |
| 23:45.54 | Naikrovek | MichaelB8225: lync has great phones, if you get polycom phones. |
| 23:46.10 | Naikrovek | but yes, it costs. |
| 23:46.10 | WIMPy | H.323 isn't only VOIP, it's avtual internet telephony. |
| 23:46.22 | Naikrovek | guess it depends on how much your phones are worth to you |
| 23:46.47 | gusto | well, but that's shit, i have to re-think my strategy now |
| 23:51.10 | MichaelB8225 | ok so, the feedback was a failing of the client software, not Asterisk? Can you suggest any other good softphones? |
| 23:51.40 | WIMPy | Erm, IIRC Zoiper has an option for echo cancelling. |
| 23:52.01 | gusto | so does ekiga, linphone and others |
| 23:52.03 | WIMPy | IAX phones are rare :-( |
| 23:52.09 | gusto | yes |
| 23:52.26 | gusto | neither ekiga, nor linphone supports it (according to wikipedia) |
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| 23:52.49 | WIMPy | There is the really ancient iaxphone. |
| 23:52.57 | MichaelB8225 | WIMPy, yeah it does, and enabled by default. Didn't do a stinkin thing |
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| 23:54.59 | gusto | maybe they should make a new RFC that would specify putting audio and video streams into h323 |
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| 23:55.24 | WIMPy | It's called IAX. |
| 23:55.34 | WIMPy | Well, sort of. |
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| 23:59.27 | gusto | hm |
| 23:59.49 | gusto | looks like IAX2 is the only protocol that can do it at the moment, right? |