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| 05:35.50 | zimmi1 | hi |
| 05:37.14 | zimmi1 | I've a caputre from a network communication between a Avaya PBX and a Asterisk PBX, the problem is when I call from a Avaya End-Point staging the one at the asterisk endpoint, he cannot hear a ringback. What is needed that the asterisk server signals a ringback? |
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| 05:45.06 | ChannelZ | well in SIP anyway you usually get a 183 Session Progress message |
| 05:46.25 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 05:47.34 | ChannelZ | Or a 180 Ringing |
| 05:48.42 | zimmi1 | its h323 |
| 05:48.47 | zimmi1 | not SIP |
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| 05:49.18 | ChannelZ | Can't help you there then |
| 05:51.30 | ChannelZ | There is in general two ways it could go though; Either the remote end indicates call progress in some way and the calling device generates its own ringing indication to the caller, or the remote end can bridge the call and generate ringing in-band |
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| 06:32.00 | v0lZy | lo |
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| 06:44.17 | ChannelZ | acks |
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| 06:49.16 | v0lZy | :D |
| 06:49.34 | v0lZy | Sup, everyone had a nice weekend? |
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| 07:14.57 | ectospasm | weekend isn't over... Monday is a holiday here. |
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| 07:25.53 | Roelt | you're an evil man ectospasm. you're making powerfull enemies |
| 07:26.44 | v0lZy | wheres 'here' ? |
| 07:27.51 | plundra | Opposite land, obviously :-) |
| 07:29.09 | Roelt | Austrialian National Flag Day, Foundation of the Republic Day in San Marino, Independence Day in Qatar or Independence Movement Day in Tunisia? :) |
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| 07:55.48 | Maliuta | ectospasm must be a canuck |
| 07:55.58 | Maliuta | aay |
| 07:56.31 | Maliuta | Roelt: National Flag day isn't a holiday here |
| 07:56.51 | Roelt | Maliuta, must be, the internet says so |
| 07:57.04 | Maliuta | Roelt: and you put in one too many "i"s |
| 07:57.36 | Maliuta | it's either Austrian or Australian ... one is a continent the other is next to Germany |
| 07:58.30 | Roelt | wasn't me, was the internet: http://www.earthcalendar.net/_php/lookup.php?mode=date&m=9&d=3&y=2012 |
| 07:58.31 | Maliuta | and here in .au there is no public holiday for "Flag Day". We have "Australia Day" on Jan 26 |
| 08:00.00 | Maliuta | Roelt: if you believe everything you read on the interwebs then I'll setup that site with cheap land on Sydney Harbour ... it's a great investment ;P |
| 08:00.12 | kikohnl | It's "Labor Day" here in the US, or will be in a few hours |
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| 08:00.36 | Maliuta | The only true Labour Day is May 1st |
| 08:01.22 | kikohnl | That's Lei Day! |
| 08:03.02 | Maliuta | http://en.wikipedia.org/wiki/International_Workers%27_Day |
| 08:03.57 | kikohnl | http://www.leiday.net/ |
| 08:07.56 | v0lZy | Agreed |
| 08:07.59 | v0lZy | May 1st! |
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| 08:15.11 | Maliuta | All good socialists/unionists should agree with me on that one. And not socialist in the McCarthiest/US sense of the word |
| 08:16.34 | indika1978 | hi, I use localchannels to dial 2 sip extensions in dial command. If one called party reject the call, ringing get stopped in both sip extensions. how can i sortout this? |
| 08:17.11 | Maliuta | indika1978: tried using a queue? |
| 08:17.21 | indika1978 | I use Dial(local/s@mysales1&local/s@mysales2,40) |
| 08:18.04 | indika1978 | once I reject the call in s@mysales1 both calls get stop ringing |
| 08:18.18 | indika1978 | How can I sortout this? |
| 08:18.48 | Maliuta | indika1978: ahh, try local/s@mysales1&&local/s@mysales2,40 |
| 08:18.50 | indika1978 | can I use OR instead of AND '&' or |
| 08:18.53 | ectospasm | Maliuta: US Labor Day is now. |
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| 08:19.14 | Maliuta | ectospasm: Americans are always wrong :P |
| 08:19.15 | [1]Tom | hello |
| 08:19.40 | [1]Tom | its because they eat lost of fastfood |
| 08:19.53 | [1]Tom | its because they eat lots of fastfood |
| 08:19.58 | Maliuta | indika1978: you also might like to try adding some other options to Dial(), I use ktrx |
| 08:20.25 | Maliuta | indika1978: as I said though, the other way is to use a queue |
| 08:21.00 | v0lZy | May 1st <3 |
| 08:22.34 | Maliuta | v0lZy: Join me in a rendition of "Solidarity Forever" ... one of my favourite drinking songs |
| 08:22.40 | [1]Tom | when will ya be able to call an astrisk box with an android any android and it will actually work ? |
| 08:22.55 | Maliuta | v0lZy: I also like the Billy Bragg version of "There is Power in a Union" |
| 08:23.17 | Maliuta | [1]Tom: ummm as of about 2 years ago |
| 08:23.28 | [1]Tom | ok |
| 08:23.42 | v0lZy | bandiera rossa! :D |
| 08:23.42 | Maliuta | [1]Tom: there are a couple of sip agents, and at least one iax agent |
| 08:23.48 | [1]Tom | i have install asterisk 11 and have tried the motif stuff |
| 08:23.51 | [1]Tom | and |
| 08:23.58 | [1]Tom | it work from windows gtalk |
| 08:24.03 | [1]Tom | but not from an android |
| 08:24.10 | [1]Tom | i get one way audio |
| 08:24.52 | [1]Tom | and i just tried a different android version and no audio |
| 08:25.10 | [1]Tom | i must be doing something really wrong |
| 08:26.21 | indika1978 | I ve used && and it wont work |
| 08:26.49 | Maliuta | [1]Tom: http://tinyurl.com/9grzlyt http://tinyurl.com/9mps6bk http://tinyurl.com/8pnbmt3 |
| 08:26.52 | indika1978 | is there any other option .. that if one local channel reject call and other still need to have ringing |
| 08:27.29 | ectospasm | indika1978: multiple Dial() statements... if the first doesn't answer, it will cascade to the next, etc. |
| 08:28.37 | Maliuta | [1]Tom: what _exactly do you want to do with you're 'droid->* connection |
| 08:28.42 | indika1978 | I need to have 2 or more local channel work with same time for one incoming call |
| 08:29.21 | ectospasm | local channels always were a hack... |
| 08:29.36 | Maliuta | indika1978: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 08:33.25 | indika1978 | Thanks but queue is needed for the second call comes to the queue, if the agents are not enough to answer the call and call get queued. My case is If I use Dial(local/s@mysales1&local/s@mysales2,40) and once one extension at mysales1 reject the call, other context mysales2 extension need to ring continuously. |
| 08:33.30 | indika1978 | How can I do this |
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| 08:50.52 | wdoekes | ~grandstream |
| 08:50.52 | infobot | somebody said grandstream was the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 08:50.59 | wdoekes | ~polycom |
| 08:50.59 | infobot | polycom is, like, The Polycom Song by dialing sip:polycom@leifmadsen.com or ISN 7659*460. Polycom phone are devices that are favoured by much of the community and range in price from under $100 and upwards. |
| 08:51.09 | wdoekes | ~spa |
| 08:51.09 | infobot | from memory, spa is the Software Publishers Association. SPA-xxxx from linksys/sipura. Health And Beauty Treatment For Old And Tired Asterisk Hackers |
| 08:52.26 | ChannelZ | indika1978: queues can be "ringall" |
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| 08:55.32 | ChannelZ | The problem lies in that (I think) you are wanting to call the agent and give them some menu of options first to accept or reject the call, and if they accept, have it bridge. But there's not a clean way to do that part of it, that I can think of at the moment. |
| 08:56.53 | Maliuta | indika1978: what ChannelZ said re "ringall" |
| 08:57.54 | Maliuta | wdoekes: I prefer cisco to polycom :P |
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| 08:58.26 | wdoekes | I've almost exclusively used the spa's, and am fond of those too indeed |
| 08:59.18 | wdoekes | I see the polycom requires separate licence keys for ldap dir support.. not nice |
| 08:59.26 | ChannelZ | indika1978: you said earlier "when I reject the call..." how are you 'rejecting' the call on one of the channels you are & dialing? |
| 09:00.05 | indika1978 | I press "reject" button on the sip phone. |
| 09:00.43 | indika1978 | Grandstreem GXP3000 or 3cx redcolour button |
| 09:00.51 | ChannelZ | And does it do the same thing if you dial the phones directly rather than through a Local channel and your dialplan? |
| 09:01.02 | Maliuta | wdoekes: the spa's are the poor mans cisco ;) |
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| 09:02.30 | ChannelZ | (ignore my previous thing about the menu, I assumed that's what you were doing with the local channels in the first place, not rejecting the call at the phone.) |
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| 09:33.13 | elliot98 | enters |
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| 09:34.04 | mathi | hi |
| 09:34.34 | elliot98 | is there anyway to increase the delay between when Asterisk sends out a packet before it's considered a lost packet and sends out another one? |
| 09:34.48 | mathi | where can I find documentation that I can print ? |
| 09:36.16 | bulkorok | mathi: there is printed doucumentary available: |
| 09:36.19 | bulkorok | ~book |
| 09:36.19 | 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 |
| 09:37.55 | kaldemar | elliot98: what packet? |
| 09:40.45 | elliot98 | kaldemar: a SIP response |
| 09:41.42 | elliot98 | kaldemar: it sends out another SIP response if the first one is not responded to...how does one increase the delay |
| 09:41.54 | mathi | bulkorok: thanks :) |
| 09:42.32 | kaldemar | elliot98: response... see timers in sip.conf. |
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| 09:44.56 | elliot98 | kaldemar: thanks |
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| 10:43.19 | kchehab | i have a nat problem since i am using asterisk 1.6 with TLS enabled and i have two interfaces one on public IP and the other on a aprivate IP,how to set both IPs with no problem i used to add externalIP and localnet and that did work |
| 10:44.20 | kchehab | kindly adivse |
| 10:45.49 | ectospasm | iirc, Asterisk can only bind to one IP, or all (0.0.0.0), but not strictly two interfaces |
| 10:46.25 | ectospasm | set bindaddr=0.0.0.0 |
| 10:46.31 | ectospasm | ...in sip.conf |
| 10:48.06 | kaldemar | "and that did work". what is the issue? |
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| 10:52.36 | indika1978 | Still I didnt get a answer for my question. If I use Dial(local/s@mysales1&local/s@mysales2,40) and once one extension at mysales1 reject the call, other context mysales2 extension need to ring continuously. How to do thia |
| 10:52.38 | indika1978 | this |
| 10:54.58 | kaldemar | you don't. |
| 10:55.40 | ectospasm | are queues not what you want, indika1978 ? |
| 10:55.47 | indika1978 | http://pastebin.com/xVx1x0hk ... pls see my dial plan ... |
| 10:57.08 | indika1978 | i have done this. and once call comes to context sales it ring 4001 and 4002. If i reject the call, it will hagup both phones .. should not happen in local channels |
| 10:57.11 | indika1978 | pls help |
| 10:58.46 | WIMPy | That should indeed not happen. Show us how it happens. |
| 10:58.53 | indika1978 | in line 6 ... it should be corrected as .....exten => s,n,Dial(local/s@mysales1&local/s@mysales2,,40) |
| 11:00.13 | indika1978 | pls refer new post ... http://pastebin.com/8SpYSDUv for updated version |
| 11:00.33 | kchehab | ectospasm i did that and tried alot even i set qualify=yes nat=yes etc.... but do asterisk 1.6 have a problem with TLS ? |
| 11:00.36 | kchehab | in nat |
| 11:07.13 | ectospasm | ALL versions of 1.6 are EOL |
| 11:09.26 | WIMPy | indika1978: You may reject the call on the phone, but that causes your dialplann actually accept the call and send it to voicemail. |
| 11:09.53 | WIMPy | If you don't want it, don't do it. |
| 11:10.55 | WIMPy | And if you want VM, don't do it for the users seperately, but after the dial to the local channels. |
| 11:13.15 | indika1978 | WIMPy, thanks. Let say A and B take the call and A need to reject it. But B wants to answer the call. How can I do that with this dialplan? |
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| 11:26.00 | indika1978 | I feel like my dial plan is too big to read. sorry for that. How could I do my expectation through this. |
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| 11:33.20 | WIMPy | I just told you. |
| 11:33.41 | WIMPy | Remove the voicemail stuff. |
| 11:34.12 | WIMPy | Or actually you could remove everything except for the Dial from mysalesX. |
| 11:34.47 | indika1978 | ok thanks I will do it |
| 11:40.00 | kchehab | how to flush/empty /var/log/asterisk/master.db: |
| 11:40.09 | kchehab | its size 10 GB |
| 11:42.57 | WIMPy | Wow. How did you get it to that size? |
| 11:43.22 | kchehab | WIMPy its running since 3 years ago |
| 11:43.29 | WIMPy | I think you can just delete the file if you don;t care about any contents. |
| 11:43.50 | WIMPy | Just running for a ling time shouldn;t make it grow indefinitely. |
| 11:44.10 | WIMPy | Otherwise there's database deltree and the like. |
| 11:44.45 | kchehab | WIMPy if i delete it ,asterisk will build a new one since it been reloaded again |
| 11:44.58 | WIMPy | Err, wait. Is that a log or did you just put the DB there? |
| 11:45.23 | kchehab | WIMPy its a DB |
| 11:45.50 | kchehab | WIMPy /var/log/asterisk/master.db |
| 11:46.46 | beebeeep | kchehab: mb you can truncate file with ':> /var/log/asterisk/master.db' and then run 'module reload cdr' from asterisk cli? |
| 11:49.41 | kchehab | beebeeep please can you specify more hopw i can run this command |
| 11:50.11 | kchehab | i ran it in the command line as :> /var/log/asterisk/master.db and waiting an answer :) |
| 11:50.31 | kchehab | beebeeep yes its empty |
| 11:50.31 | beebeeep | according to sample configs, master.db relates to sqlite cdr storage, mb you can check file type using 'file' util |
| 11:50.33 | kchehab | thanls |
| 11:50.34 | beebeeep | ah |
| 11:50.35 | kchehab | thanks |
| 11:51.12 | bulkorok | somebody got experience with patton smartnodes?! |
| 11:52.14 | kchehab | beebeeep i think you are and expert in asterisk i have a problem since i have two interfaces Public and private ,and i have anat transmitting error even i after i set externalIP= localnet= qualify=yes nat=yes |
| 11:53.43 | beebeeep | kchehab: what kind of problem? |
| 11:53.54 | beebeeep | *kind of error |
| 11:54.20 | beebeeep | did you check Contact: field in sip debug? |
| 11:56.15 | beebeeep | cd /home/miga/work/virt/var/www/well_package |
| 11:56.43 | beebeeep | oh sry folks, missed copypaste ^) |
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| 12:34.29 | _zoom_ | hi, |
| 12:35.09 | _zoom_ | is there any free software that could work with polycom video conference system |
| 12:35.10 | _zoom_ | ? |
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| 12:54.47 | indika1978 | WIMPy: at last I am with this dial plan working. But with another small issue. Let say A answered the call ,B does not answer the call. If B hangup the handset still A 's ringing stops and call is going to voice mail. How can I get this not to happen to A. means A still ringing the phone and still let A to have option to get the call. |
| 12:55.58 | indika1978 | http://pastebin.com/4agpa94t |
| 12:56.01 | WIMPy | That is the same issue. |
| 12:56.07 | WIMPy | Exactely the same. |
| 12:56.47 | indika1978 | I ve change the dial plan. |
| 12:57.25 | indika1978 | according to your instructions and alter some to simple the logic |
| 12:57.33 | indika1978 | pls just have a nap |
| 12:58.27 | WIMPy | Ok, better, but you seen logically, you didn;t remove the voicemail stuff, you just wrote it in a different way. |
| 12:58.51 | WIMPy | Get rid of the Goto()s after the Dial() in mysalesX. |
| 12:59.15 | indika1978 | yes I ve tried and it happens same thing |
| 12:59.26 | WIMPy | If you want Voicemail, do it after the Dial() in sales. |
| 12:59.51 | indika1978 | yes I followed your instructions exactly |
| 12:59.57 | indika1978 | and did it . |
| 13:00.06 | indika1978 | wait I again post the ouput |
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| 13:01.56 | indika1978 | http://pastebin.com/3LvDPkVh new one without voice mail or goto voicemail |
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| 13:03.58 | indika1978 | A and B ringing. A answered. B still rings. A hangup the call. B stops ringing. .----<--- i need B still ring and able to get the call |
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| 13:14.52 | indika1978 | Sorry to bother you all, does it have any solution? |
| 13:17.51 | WIMPy | Busy doing other stuff. |
| 13:18.19 | WIMPy | When exactely does the other phone stop ringing? |
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| 13:28.34 | ACiDV | Hi, anyone can give me hints on how to continue dialplan execution when someone hangup from a ConfBridge application (Asterisk 10 and 11) ? When the last party leave, I must execute some command on the "Hangup" dialplan |
| 13:30.20 | [TK]D-Fender | ACiDV: "h" Asterisk Standard Extension |
| 13:30.56 | *** join/#asterisk soulslayer (~soulslaye@213.145.99.82) |
| 13:31.23 | ACiDV | [TK]D-Fender: I've tried but it do nothing :( The only fix I see for now is to trap the Hangup event from AMI and execute task from an external script, not tested yet, it my next experiment |
| 13:31.51 | [TK]D-Fender | ACiDV: Show me it doing "nothing" and the dialplan to match |
| 13:32.47 | soulslayer | hi there guys :) I'm trying to execute diferent scripts if user answer a certain call and if call is not answered eg.(declined left to expire or whatever) i use ${DIALSTATUS} for that but it's always 'UNKNOWN' obviously something in my conf is wrong can someone provide any guidelines about that ? |
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| 13:35.00 | indika1978 | http://paste2.org/p/2185170 ........ clearly there in the post |
| 13:35.27 | WIMPy | Whow do yu do that? I've never seen UNKNOWN. According to core show... that's not even possible. |
| 13:35.43 | soulslayer | <PROTECTED> |
| 13:36.17 | kaldemar | soulslayer: that's not DIALSTATUS. |
| 13:36.22 | WIMPy | That's the channel status, not DIALSTATUS. |
| 13:36.26 | soulslayer | yep ... that was the other option |
| 13:36.32 | soulslayer | i see |
| 13:36.49 | soulslayer | sorry then my question is wrong |
| 13:37.43 | soulslayer | where can i read about handling diferent dialstauts ... i try to dig in the docs but the data is splited ... and ... cant find the exact issue |
| 13:37.43 | soulslayer | ? |
| 13:38.44 | WIMPy | What exactely do you want to do? |
| 13:38.53 | WIMPy | And when. |
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| 13:39.15 | soulslayer | i want to execute one script when the call is answered from the called sip and other script if it's not answered |
| 13:39.35 | soulslayer | by not answered i talk about missed declined or anything else |
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| 13:39.51 | WIMPy | Then you need to check ${DIALSTATUS} and possibly ${HANGUPCAUSE}. |
| 13:39.58 | [TK]D-Fender | soulslayer: What will the script do? |
| 13:40.25 | soulslayer | it will notfy webbackend that the call is picked or not picked |
| 13:40.31 | indika1978 | guys you missed me... :( |
| 13:40.44 | indika1978 | misssed my post |
| 13:40.55 | WIMPy | indika1978: I see that it says mysales2 answered, But I don't see why that happens. |
| 13:41.06 | soulslayer | the end goal is to have ... list of pending calls and when a call is picked by user it will disapear from the list if it's not pick it will become red |
| 13:41.11 | [TK]D-Fender | soulslayer: just the Dial + M() to send the ACK immediately for answered, and the next priority for all other failures |
| 13:41.45 | indika1978 | What can i do now ..? |
| 13:42.59 | soulslayer | [TK]D-Fender, you mean to pass System("execute script") as arugment on the dial application ? |
| 13:43.07 | soulslayer | sorry im not sure that i understand you completely |
| 13:43.16 | indika1978 | http://paste2.org/p/2185205 after all calls hangup |
| 13:43.20 | [TK]D-Fender | Dial + M() <--------- |
| 13:43.56 | indika1978 | is it for me ? |
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| 13:45.33 | soulslayer | ahaaaa |
| 13:45.49 | soulslayer | M() is predefined macro where i am supposed to put my System() calls |
| 13:46.28 | soulslayer | okay ... i will try that |
| 13:46.49 | soulslayer | thanks once again first grade of help |
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| 13:52.15 | WIMPy | indika1978: It's the macro. It always ends the same way. You don;t really have an option there to not accept the call. |
| 13:54.11 | indika1978 | I found the reason .. but i dont konw how to sort out that.. pls seee this post ... |
| 13:55.51 | WIMPy | I'm not really sure about that thing, either. i haven't tried such an option myself so far, but I think you need to set MACRO_RESULT to ABORT. However, I'd expect a hangup to do the same. |
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| 13:59.08 | indika1978 | ok |
| 13:59.31 | indika1978 | that means if call get hang up how can we catch that in dial plan |
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| 14:00.43 | WIMPy | In the h extension. |
| 14:01.22 | WIMPy | Or you could probably just set MACRO_RESULT to ABORT at the beginning and reset it only to accept the call. |
| 14:01.41 | WIMPy | Maybe someone who has used such a thing has a better idea. |
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| 14:20.41 | StaRetji | folks, I need help to stop asterisk do a fake ring |
| 14:20.53 | StaRetji | I am using a2b and I removed R and r |
| 14:20.56 | StaRetji | to no avail |
| 14:21.03 | StaRetji | where else I can look? |
| 14:21.22 | WIMPy | How do you receive the call? |
| 14:21.30 | indika1978 | Or you could probably just set MACRO_RESULT to ABORT at the beginning and reset it only to accept the call. ---- this biging means where ? in the macro? |
| 14:21.34 | indika1978 | pls tell me |
| 14:21.56 | WIMPy | indika1978: In the macro, yes. |
| 14:22.30 | [TK]D-Fender | StaRetji: If you didnt specify "r" then Asterisk is not responsible for "fake" ringing |
| 14:24.47 | StaRetji | [TK]D-Fender: thx, so it must be equipment then, gsm modem at the end |
| 14:24.53 | indika1978 | ok I will try to sort out.. thaks for the great help |
| 14:28.25 | WIMPy | Does your dialplan answer the call before $user does? |
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| 14:47.47 | defswork | anyone rate a good outdoor sip keypad intercom ? |
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| 15:10.45 | waxstone_ | hi all! having issues registering softclients sif phones. sip show users shows me users correctly. sip show peers shows no device registered(no host ips). tried various clients (blinl,twinkle,3cxphone) without success. i can shh to server box no problem |
| 15:11.30 | defswork | sip proxy settings ? |
| 15:11.33 | WIMPy | Firewall? |
| 15:12.31 | waxstone_ | im doing this test at home. server=box1 , clients on box2 & box3 all behind router is this an issue? because i can ssh i assume not. im really a neubie at asterisk |
| 15:13.02 | defswork | waxstone_, most likely firewall allows ssh and not sip (or subsequently rtp) |
| 15:13.39 | waxstone_ | defswork: firewall on router correct? must I open a port? |
| 15:14.06 | defswork | sorry - all machines are at home ? |
| 15:14.12 | waxstone_ | yes |
| 15:14.20 | defswork | no - no firewall then |
| 15:14.33 | waxstone_ | ok |
| 15:14.37 | defswork | connecting anything over sip ok ?> |
| 15:14.47 | waxstone_ | nothing |
| 15:15.13 | waxstone_ | ive followed the online manual from asterisk.org exactly |
| 15:15.22 | defswork | what sip client ? |
| 15:15.26 | waxstone_ | server seems to be setup correctly |
| 15:15.47 | waxstone_ | blink, twinkle & 3cxphone |
| 15:15.58 | waxstone_ | none register correctly |
| 15:16.06 | defswork | correctly or at all ? |
| 15:16.12 | waxstone_ | at all |
| 15:16.51 | defswork | you've probably set them up with wrong settings |
| 15:17.07 | defswork | and turn on sip debug in * console |
| 15:17.25 | defswork | and try linphone - it's only 3-4 fields to configure |
| 15:17.26 | WIMPy | Or an additional firewall. |
| 15:17.33 | defswork | WIMPy, hes at home |
| 15:17.42 | defswork | all on same lan |
| 15:17.43 | WIMPy | 'sip set debug on' and see if you receive anything at all. |
| 15:17.49 | waxstone_ | core set verbrose 9... & sip set debug on |
| 15:17.52 | waxstone_ | nothing |
| 15:18.14 | waxstone_ | sip show peers nothing under Host(Unspecified) |
| 15:18.45 | waxstone_ | ill google this linphone thanks |
| 15:18.55 | WIMPy | Then either your client doesn't know how to reach the server or you have some extra firewall running on at least one of the macoines. |
| 15:19.10 | StaRetji | [TK]D-Fender: sorry for late ping, but I still strugle. I placed equipment on hardware switch and no fake ring. So I went back checking a2b agi.conf, no R or r, but asterisk still rings SIP/blablah is ringing |
| 15:19.19 | waxstone_ | server is linux with selinux off |
| 15:19.47 | StaRetji | I'm going crazy, don't know where else to look :/ |
| 15:19.47 | WIMPy | iptables? |
| 15:20.19 | [TK]D-Fender | StaRetji: If * hasn't answered then the endpoint has passed on a indication of ringing and * merely relays that and you other end is generating tone |
| 15:20.23 | waxstone_ | WIMPy: i dk much about editing iptables. could this be the issue? |
| 15:20.26 | WIMPy | StaRetji: Where are you getting the calls from? And are you answering the call before $user answers? |
| 15:20.35 | [TK]D-Fender | If * answered the incoming first then it generates audio for the outgoing. |
| 15:20.39 | [TK]D-Fender | but it is not fake" |
| 15:20.52 | WIMPy | waxstone_: If you have them set up, they might be the issue. |
| 15:21.21 | StaRetji | I have problem, customer has server, calls my server and imidiatelly heres US ringing, instead of EU ringing, so I assume it is fake |
| 15:21.48 | waxstone_ | WIMPy,defswork: ok ill do some research into these thanks all for the leads ill be back later |
| 15:21.51 | StaRetji | he claims other routes are ok, only my route/* |
| 15:22.01 | [TK]D-Fender | StaRetji: And I jsut told you how to determine which device is generating it. |
| 15:22.06 | [TK]D-Fender | and it is NOT "fak" |
| 15:22.19 | WIMPy | StaRetji: Who of you has eu and whu has us tonezone configured? |
| 15:22.44 | [TK]D-Fender | the device that generates it is doing so based on a REGION setting. Ring is local to the device that generates it. It is clearly not coming in as raw inbound audio. |
| 15:22.57 | StaRetji | I have gsm modem |
| 15:23.01 | StaRetji | as a trunk |
| 15:23.14 | StaRetji | and my * is set for outbound calls via this modem |
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| 15:23.31 | StaRetji | customer is peer |
| 15:23.45 | WIMPy | That will surely not generate rinback. So it looks as if your dialplan answers the call immediately. That is not a sensible idea, usually. |
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| 15:24.27 | [TK]D-Fender | StaRetji: I've answered this already. there is no more to say. Go look and prove which end is generating it. It isn't fake, its just a n INDICATED state, not inbound audio. |
| 15:24.28 | WIMPy | So it is a sip peer dialling out via the GSM who hears the wrong kind of ringing? |
| 15:24.29 | StaRetji | he says, that he can't faltrough because of this and not goint to use my route, so I bang my head |
| 15:25.09 | StaRetji | WIMPy: he imidiatelly hears ringing, though GSM still not doing anything |
| 15:25.32 | StaRetji | [TK]D-Fender: thx dude, I try to understand all |
| 15:25.38 | WIMPy | Sounds like he needs to configure his phone. |
| 15:25.58 | [TK]D-Fender | Sounds like he needs to prove is one device or * itself DID answer and actually LOOK at the complete call |
| 15:26.43 | WIMPy | Depending on how you dial on a progress message might be misunderstood. |
| 15:26.45 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.wanadoo.nl) |
| 15:28.01 | Tagor | I upgraded Asterisk (1.6.2.9-2) but suddenly the Monitor command is not working anymore. I get no errors, there's enough diskspace, but somehow it does not create a recording. |
| 15:28.06 | Tagor | Anyone know what's wrong? |
| 15:28.31 | Tagor | During the call it also doesn't create the two files which it normally did |
| 15:28.57 | StaRetji | WIMPy: tonezone, where can I set this? |
| 15:30.16 | WIMPy | In the channel configs, in the dialplan (language) and the default in asterisk.conf. |
| 15:30.30 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 15:30.35 | StaRetji | hm, I am actually desperate here, but I realize, I don't need no timezone, I just need asterisk not to pass any tones at all, guess htat is not it, right |
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| 15:32.07 | WIMPy | You need to find out where it is generated. For SIP that's usually in the phone itself. |
| 15:32.31 | StaRetji | here it is wholesale customer |
| 15:33.01 | [TK]D-Fender | StaRetji: Stop asking where to set it. You haven't proved precisely WHICH step is generating it. |
| 15:33.02 | StaRetji | has it's own server, and he claims I generate us ring tone, he advised me to remove r and R from a2b agi.conf, which I did |
| 15:33.14 | StaRetji | well, I did |
| 15:33.18 | StaRetji | I turned of gsm mode |
| 15:33.20 | StaRetji | modem |
| 15:33.33 | [TK]D-Fender | StaRetji: We do not see this. |
| 15:33.33 | StaRetji | and * rings before actually fails to make a call |
| 15:34.14 | WIMPy | You clearly have somethign stange going on there. |
| 15:34.14 | StaRetji | thanks for patiance, I guess it is hard to explain to a noob |
| 15:34.21 | WIMPy | So back to... |
| 15:34.45 | WIMPy | Are you answering the call in your dialplan before $user answers? |
| 15:34.59 | WIMPy | SSurely sounds like you are. |
| 15:35.42 | [TK]D-Fender | Nothing sounds like anything. We have no debug. |
| 15:35.59 | StaRetji | well, not sure, I look at extensions.conf, nothing much there, only a2b lines |
| 15:36.15 | [TK]D-Fender | StaRetji: LOOK.; AT. THE. CALL. |
| 15:36.38 | StaRetji | thought it might be option in agi.conf answer_call=no |
| 15:36.48 | StaRetji | ok, got it, will try to do SIP Debug |
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| 15:36.54 | StaRetji | but what to LOOK AT? :) |
| 15:37.08 | WIMPy | The verbose output |
| 15:37.22 | [TK]D-Fender | all debug |
| 15:37.42 | [TK]D-Fender | CLI Verbose, SIP, EVERYTHING |
| 15:37.55 | carrar | ORLD YES |
| 15:37.58 | carrar | WORLD YES |
| 15:38.15 | StaRetji | ok folks, very clear, look everything |
| 15:38.18 | StaRetji | wish me luck! |
| 15:38.26 | carrar | wishing you more then luck |
| 15:38.57 | StaRetji | thx carrar :) |
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| 16:04.15 | xoveruk | hi |
| 16:04.35 | xoveruk | is it possible to set PINs on IAX and SIP extensions to lock down a phone? |
| 16:05.10 | WIMPy | Write a piece of dialplan. |
| 16:05.26 | WIMPy | That's how it's done. |
| 16:05.56 | xoveruk | what do you mean? |
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| 16:06.35 | WIMPy | vi extensions.conf :-) |
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| 16:08.44 | xoveruk | I have codes on the dialplans, however I am after a per user code |
| 16:08.59 | xoveruk | rather than per route |
| 16:09.03 | [TK]D-Fender | xoveruk: Your dialplan does whatever you tell iut to. |
| 16:09.28 | [TK]D-Fender | Point your peers to whatever context you want and you tell it what to allow and what not ask for auth for. |
| 16:10.34 | xoveruk | OK but can it prompt for auth per extension? |
| 16:10.42 | xoveruk | or per context? |
| 16:10.57 | WIMPy | What ever you want. |
| 16:11.13 | xoveruk | where can i find indepth documention on this? |
| 16:11.31 | WIMPy | ~book |
| 16:11.31 | 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 |
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| 16:11.51 | xoveruk | cool |
| 16:11.51 | [TK]D-Fender | xoveruk: You want ONE phone treated differently? Point it to a different context |
| 16:11.54 | xoveruk | thanks |
| 16:12.05 | [TK]D-Fender | or check who the caller is. |
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| 16:12.11 | [TK]D-Fender | Channel name, callerID, etc |
| 16:12.18 | [TK]D-Fender | heads off for the afternoon |
| 16:12.21 | xoveruk | Got you. |
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| 16:13.41 | navaismo | moorning |
| 16:29.30 | xoveruk | Is it therefore not possible to set a PIN on a phone? |
| 16:29.49 | xoveruk | to prevent another user from using someone elses phone to dial internationally for example. |
| 16:32.11 | pabelanger | not sure, you'd have to see if that phone supported it. It has nothing to do with asterisk |
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| 16:42.02 | mechbangirc | hello guys, I want to add an extension where my asterisk server can scan a page through an attached usb scanner. whats the best way to do it? |
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| 16:48.29 | navaismo | mechbangirc, not sure how you access to your scanner but yu can use the system command or an agi script |
| 16:51.41 | mechbangirc | navaismo, the command is something like this: `scanimage --format tiff --resolution $resolution --mode $mode > $tiffname` the problem is that it works directly in bash perfectly. but not with system() app in dialplan. i suspect the problem is output redirection. not sure though |
| 16:52.14 | WIMPy | Put it in a script. |
| 16:52.43 | mechbangirc | thats exactly what I did, this was just the core line of my script |
| 16:53.10 | mechbangirc | I want to run it in background so I can hangup the phone and process still continues |
| 16:53.12 | WIMPy | Permissions? |
| 16:53.51 | mechbangirc | I've run the script as asterisk user in bash and it still works but not with dialplan |
| 16:54.23 | WIMPy | The environment will certainly be different. |
| 16:54.29 | navaismo | you run your script as root in the bash? The user of asterisk is other than root? |
| 16:55.30 | mechbangirc | no, I changed the user to asterisk with `su asterisk`, i'm pretty sure its not permission issue. it even creates a tiff file but 0 bytes |
| 16:55.31 | navaismo | as WIMPy said: check the permissions |
| 16:55.42 | *** part/#asterisk xoveruk (~rich@host31-53-45-187.range31-53.btcentralplus.com) |
| 16:56.16 | mechbangirc | is agi() better way to do this instead of system()? |
| 16:56.36 | WIMPy | No |
| 16:56.55 | WIMPy | AGI only makes sense if you want to do something with the call from the script. |
| 16:57.15 | mechbangirc | yes, thats why I didn't try it |
| 16:57.57 | WIMPy | Try to redirect the output to a file for debugging. |
| 16:59.34 | madduck | asterisk is logged in to sipgate.de and I have voicemail configured for incoming messages. Unfortunately, the audio channel seems to die after a couple of seconds so the voicemail greeting just goes silent |
| 16:59.45 | madduck | any idea what may be the cause of this? |
| 17:01.31 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
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| 17:04.13 | WIMPy | Can you make Asterisk send something else than what's configured as host= as the domain in the to: line of an invite. |
| 17:05.28 | madduck | there is nothing in the logs :( |
| 17:06.48 | WIMPy | Did you turn up verbose and debug? |
| 17:06.56 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:08.08 | madduck | core set verbose|debug 100 ? |
| 17:08.28 | WIMPy | 9 should be planty |
| 17:08.39 | madduck | nothing… |
| 17:08.40 | WIMPy | Not sure if there's anything >4. |
| 17:09.04 | WIMPy | But it works for a short while? |
| 17:09.11 | madduck | yes |
| 17:09.31 | WIMPy | Must be some rather strange networking issue. |
| 17:09.52 | madduck | it's only for voicemail though, never for a normal call |
| 17:10.19 | WIMPy | Do you Answer() the call before you send it to VoiceMail()? |
| 17:11.32 | madduck | i basically use an extended version of stdexten |
| 17:11.50 | madduck | it does not Answer() |
| 17:11.54 | madduck | however, I can hear the beginning just fine |
| 17:12.09 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:13.24 | pabelanger | nothing is above 10 for debug and verbose |
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| 17:19.32 | WIMPy | I can't remember why, but I needed an Answer() before VoiceMail(). |
| 17:21.23 | madduck | well, here it Dial()'s a SIP number to let my phone ring and after a timeout, it uses ${DIALSTATUS} to go to Voicemail. |
| 17:21.38 | madduck | http://slexy.org/raw/s21uDgnXeu |
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| 17:24.10 | waxstone | WIMPy: CentOS iptables were the issue, thanks again for your help. sip clients successfully call each other now |
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| 17:45.45 | cusco | hi |
| 17:45.49 | cusco | not sure if it makes sense but... |
| 17:46.10 | cusco | is it possible to know the MAC addr of registered peer? ? |
| 17:48.42 | navaismo | cusco, if you have the IP you can try to scan the mac with arp or nmap or something |
| 17:48.45 | WIMPy | Only if it is on the LAN or thells you in some header. |
| 17:49.01 | cusco | ok... |
| 17:49.11 | *** join/#asterisk brian98 (~brian98@79.97.24.23) |
| 17:49.24 | cusco | well thing is.. we have about 100 peers at one office.. |
| 17:49.42 | cusco | and they all have softphones configured with their usernames |
| 17:50.03 | cusco | so.. someone connected couple of nights ago, and started calling other extensions making jokes and all |
| 17:50.35 | cusco | and all I have is: 00:13:72:C9:8F:7E |
| 17:50.37 | cusco | oops |
| 17:50.39 | cusco | chan_sip.c: -- Registered SIP '690' at 10.67.164.111 port 37389 |
| 17:50.57 | cusco | useragent was '3cx for android' |
| 17:51.11 | jpsharp | Well, cross reference that IP with your DHCP server. |
| 17:51.12 | WIMPy | arp |
| 17:51.17 | cusco | its not there |
| 17:51.20 | cusco | now |
| 17:51.28 | cusco | that was on the weekend |
| 17:51.30 | cusco | lol |
| 17:51.37 | cusco | I'm looking at the arp table |
| 17:51.49 | cusco | we do not offer dhcp in 10.67 |
| 17:52.03 | jpsharp | You're pretty boned, then. |
| 17:52.04 | WIMPy | Buy who knows the password for 690? |
| 17:52.13 | WIMPy | Why do you accept traffic from there then? |
| 17:52.31 | cusco | anyone with access to the computer that has 690 configured in softphone? |
| 17:52.51 | cusco | hmm |
| 17:52.53 | WIMPy | That's more than one user? |
| 17:52.56 | cusco | yes |
| 17:53.01 | cusco | we rotate computers |
| 17:53.14 | WIMPy | Why du multiple users share one sip account? |
| 17:53.39 | WIMPy | Don't the PCs have users? |
| 17:53.59 | cusco | 3, like 'telemarketing', 'tecnician' 'sencond-line' |
| 17:54.01 | cusco | lol |
| 17:54.23 | cusco | yea.. well I could go on and answer but thats over the point.. we will change that in the future.. its planned already |
| 17:58.42 | *** join/#asterisk boch (~boch@200.107.200.9) |
| 17:58.45 | boch | hi all |
| 17:58.55 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:59.27 | boch | could someone give me a hand? im having problems to use dahdi, im getting WARNING[3848]: chan_dahdi.c:5191 dahdi_confmute: DAHDI confmute(0) failed on channel 61: Invalid argument |
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| 18:22.15 | ChannelZ | channel 61 is the spanish channel and really does need to be muted. |
| 18:22.44 | TriJetScud | That's racist |
| 18:22.58 | navaismo | :( |
| 18:23.19 | WIMPy | Yes, let's stop the bad jokes and bring on the racist ones :-) |
| 18:23.22 | ChannelZ | Why? Gringos speaking bad spanish is just as annoying |
| 18:23.55 | TriJetScud | Racist too. :p |
| 18:24.06 | ChannelZ | You must be a liberal |
| 18:24.12 | ChannelZ | Everyone's a racist! |
| 18:24.49 | TriJetScud | No you're racist ;} |
| 18:25.29 | TriJetScud | Get the joke already? |
| 18:26.28 | ChannelZ | No because race hasn't even been mentioned. |
| 18:26.53 | ChannelZ | So unless the joke is that "the joke makes no sense"... |
| 18:26.59 | TriJetScud | Haha |
| 18:27.10 | WIMPy | You're a linguist. |
| 18:27.17 | boch | ChannelZ, why so sad? why the rancor |
| 18:28.23 | ChannelZ | I'm working on what should be a day off |
| 18:28.48 | WIMPy | Then you didn't work hard enough. |
| 18:28.56 | ChannelZ | Pretty much |
| 18:29.11 | ChannelZ | I'm at least doing it at home in my underwear so it's not all bad |
| 18:29.17 | WIMPy | hopes someone gets that joke. |
| 18:32.06 | jpsharp | I'm not racist. I hate everyone equally. |
| 18:33.50 | ChannelZ | See, there we go. High-five! |
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| 18:54.34 | TriJetScud | jpsharp can commit suicide then lolol |
| 18:56.15 | jpsharp | Why thank you. |
| 19:04.58 | SeRi | I am Latino and I hate the spanish channels. |
| 19:05.14 | SeRi | except the big chichis :) |
| 19:07.10 | navaismo | ¬¬ |
| 19:07.41 | SeRi | lol |
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| 19:09.54 | ChannelZ | Telenovelas are pretty horrendous |
| 19:10.01 | ChannelZ | Not that english ones are much better |
| 19:10.16 | SeRi | ChannelZ: Thats what makes me hate the spanish channels |
| 19:10.32 | SeRi | My wife glues on to the TV. |
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| 19:43.17 | Serious_Sam | Hi there! Please advice some good books about telephony. |
| 19:46.07 | sruffell | ~book |
| 19:46.07 | 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 |
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| 19:56.33 | WIMPy | Did that fit the question? |
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| 20:08.50 | ChannelZ | I was going to say The Phonebook |
| 20:09.23 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 20:10.06 | WIMPy | Too obvious. |
| 20:14.08 | Serious_Sam | Thanks for the book |
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| 21:37.27 | mathi | hi |
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| 21:38.02 | mathi | in the book it is written "The DNIS number and the DID do not have to match, but typically they will." I don't understand in which case they can differ |
| 21:40.59 | WIMPy | Forwarding |
| 21:41.06 | gusto | what is DNIS number? |
| 21:41.27 | gusto | hi WIMPy |
| 21:41.30 | gusto | what's new? |
| 21:42.01 | mathi | forwarding what to where? |
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| 21:42.52 | WIMPy | A call, somewhere. |
| 21:42.59 | fenrus | a->b->c :) |
| 21:44.43 | mathi | I don't think that answers my question, here it's only about accepting calls. the DID would be the number to dial to enter the PBX, and the DNIS allows me to see the number that has been called |
| 21:45.11 | WIMPy | Probably. |
| 21:45.25 | WIMPy | I'm never really sure what ends up where in Asterisk. |
| 21:46.25 | WIMPy | But I did notice that it's sometime ambiguous |
| 21:47.23 | gusto | ah |
| 21:47.32 | gusto | depends on how complex your dialplan is |
| 21:48.11 | WIMPy | ? |
| 21:49.25 | mathi | I'm still a noob in telephony, but it seems that if someone calls to my Asterisk PBX, and I forward him to another external number, I have to pay that last call ? :-( |
| 21:49.47 | WIMPy | Sure. |
| 21:49.50 | carrar | mathi, it means you can order a 10 digit DID, but have them only send like the last 4 digits |
| 21:49.54 | WIMPy | Who else? |
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| 21:50.02 | carrar | all done at the PRI carrier side |
| 21:50.15 | carrar | some PBX's receive 4 digits from Telco's |
| 21:50.18 | WIMPy | So it has nothing to do with forwarding? |
| 21:50.20 | carrar | instead of all 10 |
| 21:50.34 | carrar | so in that case |
| 21:50.38 | carrar | they don't "amtch" |
| 21:50.40 | WIMPy | The whole caller ID stuff really needs some documentation. |
| 21:50.43 | carrar | match |
| 21:50.47 | mathi | carrar, aahhh got it thanks |
| 21:50.50 | carrar | with regard to DNIS/DID |
| 21:51.27 | SeRi | anybody using any external FXS? |
| 21:51.28 | carrar | come telco's can trasnlate the DID to something else |
| 21:51.30 | carrar | some |
| 21:51.43 | SeRi | FXS/FXO |
| 21:51.58 | mathi | WIMPy, back to my last uqestion, i cannot make the caller call directly the external number, and my PBX would not handle it. I guess that's impossible:) I do'nt want to pay |
| 21:52.57 | WIMPy | I'm not sure what you want to say. |
| 21:53.11 | carrar | HAMMER TIME |
| 21:53.48 | *** join/#asterisk gociii (~gociii@2.29.138.97) |
| 21:54.00 | gociii | hi ppl |
| 21:54.23 | mathi | carrar, about DID/DNIS.... if I have extensions for each of the DID to handle, how is the DNIS even useful ? As I know the number that has been called as I have an extension for it ... |
| 21:55.00 | carrar | DNIS is usefull so you know what number was dialed |
| 21:55.08 | carrar | regardless how many digits it might be |
| 21:55.39 | carrar | but some PBX admins are lazy |
| 21:55.39 | mathi | carrar, but if I have this: exten => 4165551234,1,SomeApplication() I know the number 4165551234 has been called, why do I need DNIS ? |
| 21:56.00 | carrar | and only want to match the extensions directly to DID's so they have the telco cutt off the left 6 digits |
| 21:56.11 | carrar | if they use 4 digit extensions |
| 21:56.17 | carrar | and also if they happen to match up |
| 21:56.51 | carrar | mathi, if you didn't havce DNIS, 4165551234 would not match at all |
| 21:57.28 | carrar | it would just ring and you would have to answer it, not knowing what number they dialed |
| 21:57.39 | gociii | hi guys… a newbie here… would like to develop on asterisk and would require some insight in regards tools to use. My target is to run DSP checks on audio from a SIP channel and direct asterisk in doing things accordingly… i.e. hang up or take caller to another level… my questions are: which tools would you use for development i.e. IDE and extras please? |
| 21:58.49 | gusto | what is 41 for a number? |
| 21:59.15 | mathi | carrar, so why do providers cut numbers? we would just have one number and not this DID/DNIS thing .... |
| 21:59.38 | carrar | if you have extensions 4000-4100 |
| 21:59.39 | carrar | and |
| 21:59.53 | carrar | you own 206-812-4000 - 206-812-4100 |
| 22:00.05 | carrar | You could have your tleco just send the last 4 digits |
| 22:00.08 | gociii | any asterisk developer gurus in the house please? |
| 22:00.20 | carrar | and you would not have to add the full 10 digits |
| 22:00.44 | carrar | this was more usefull on older P BX, where you had to enter in digits 1 by 1 |
| 22:00.58 | carrar | so just a 1:1 match made life easy |
| 22:01.03 | navaismo | gociii, there is an irc developer channel asterisk-dev |
| 22:01.16 | gociii | many thanks navaismo |
| 22:01.18 | gociii | :) |
| 22:01.31 | mathi | carrar, ah so external peers would have to call 206-812-4000 and internal peers would have to call 4000, and then they use the same extension |
| 22:01.50 | carrar | they would/could reach the same extension yes |
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| 22:05.13 | mathi | WIMPy, what I said earlier was that when someone calls from PSTN and i want to redirect him to another external phone through PSTN, we get: caller -> my Asterisk PBX -> receiver. And I would have to pay the call PBX -> receiver. Isn't it possible for my PBX to make the caller call the receiver and end the call throught he PBX |
| 22:06.12 | WIMPy | No. |
| 22:06.31 | mathi | come on WIMPy we live in 2012 |
| 22:06.43 | WIMPy | You may be able to use deflection or transfer to get the channels free, but that won;t change the billig. |
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| 22:07.18 | WIMPy | Or you send some app to the caller that does a little search&replace on his phone. |
| 22:07.19 | WIMPy | :-) |
| 22:08.24 | mathi | but it is possible to call for free through the internet? |
| 22:08.48 | WIMPy | Let's say without paying for the call. |
| 22:08.52 | mathi | from my PBX to an external phone |
| 22:09.06 | mathi | what are the possible ways to achieve this ? |
| 22:09.14 | WIMPy | If you can reach it that way. |
| 22:09.35 | WIMPy | Basically you call an URL instead of a number. |
| 22:12.37 | mathi | I think there are also other solutions like Zoiper |
| 22:13.11 | WIMPy | Yes, you can use your PC as phone. |
| 22:13.45 | mathi | and a physical SIP phone, would use an URL? |
| 22:13.53 | WIMPy | Yes |
| 22:14.03 | WIMPy | Even if you just enter digits. |
| 22:17.40 | mathi | well most of the people still phone via PSTN and pay. the reason is that SIP is not stable yet? |
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