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| 01:13.49 | voip_troll | Is there any way for asterisk hairpin audio for 2 calls from/to PSTN (via SIP trunk)? Currently a re-invite is happening, but I'd prefer Asterisk handles all the media. |
| 01:14.28 | voip_troll | Running 1.6.2.11 |
| 01:16.37 | WIMPy | canreinvite/directrtpsetup |
| 01:19.27 | voip_troll | WIMPy: Thanks - I had been setting it for the users but not the trunk. |
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| 01:34.26 | *** join/#asterisk sweetpi (~sweetpi@unaffiliated/sweetpi) |
| 01:40.28 | sweetpi | What would it be called if I need someone to be able to call the asterisk box, enter a security code and phone number then have the asterisk box 3-way call the supplied phone number? |
| 01:46.18 | p3nguin | If you call Asterisk and Asterisk calls another number, that's only a two-way call. You could use DISA for that, but I'm sure it could be done just as easily with Authenticate(), Read() and Dial(). |
| 01:47.37 | WIMPy | call through |
| 01:49.53 | sweetpi | yes call through. thank you |
| 01:50.32 | sweetpi | p3nguin: I didnt know what to call it. there is only one line, so it needs to do it that way |
| 01:51.02 | p3nguin | One line? Like an analog phone line? |
| 01:51.06 | sweetpi | yes |
| 01:51.56 | p3nguin | I think I understand why you said 3-way call, then. |
| 01:52.19 | p3nguin | Asterisk is actually a "phone" in the call. |
| 01:53.02 | p3nguin | Normally it would bridge the two other actual phones and not be a party in the call. |
| 01:54.50 | sweetpi | Yes, I dont want asterisk to be part of the call, but I didnt know if that was possible with one analog line |
| 01:55.22 | sweetpi | Ive never used asterisk before, do you think it is overkill for what im trying to do? |
| 01:55.56 | p3nguin | What's the point of it? |
| 01:56.53 | p3nguin | It might be possible, but I wouldn't know how to configure it that way. I don't see the purpose. If you can make a call out to Asterisk, you could have just called the other person directly and not bothered with Asterisk at all. |
| 01:57.43 | sweetpi | To let my friends kids be able to call their dad for free. |
| 01:58.20 | sweetpi | without me having to manually do it and be around when they want to call |
| 01:58.44 | p3nguin | Okay, that actually makes sense. |
| 02:00.00 | sweetpi | so call through is what I need to read about then? |
| 02:00.27 | p3nguin | How are you connecting Asterisk to your phone line? |
| 02:00.52 | p3nguin | What type of interface/tech? |
| 02:01.42 | sweetpi | I was hoping I could just add a old modem to a box I have sitting around. or not? |
| 02:01.53 | p3nguin | Not likely. |
| 02:02.45 | p3nguin | ~modems |
| 02:02.45 | infobot | rumour has it, modems is something you can not use as an fxo interface under asterisk. See http://www.soft-switch.org/cards.html#modems |
| 02:03.09 | sweetpi | hmm. so even if it worked on linux it still might not work :/ |
| 02:05.37 | sweetpi | yeah doesnt look promising. what kind of cost am I looking at for the cheapest supported card? |
| 02:06.25 | p3nguin | You'd probably be better off buying an ATA that can deal with 3-way calling appropriately (assuming one exists). |
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| 03:01.48 | sweetpi | p3nguin: I'm confused, it seems an ATA is to connect a phone or fax and go out on ethernet for sip? |
| 03:02.26 | p3nguin | Some ATAs have FXO ports for connecting to a phone line (wall jack). |
| 03:02.55 | *** join/#asterisk myster (~myster@74.198.148.18) |
| 03:02.59 | p3nguin | The SPA-3102, for example, has both FXS and FXO ports. |
| 03:03.36 | p3nguin | I cannot remember what model number has only an FXO port. |
| 03:04.55 | *** join/#asterisk crab (ams@disorder.primate.net) |
| 03:05.00 | crab | hi. |
| 03:06.43 | sweetpi | how about this? http://www.digiumcards.com/zoom_telephonics_5801_FXO_FXS.html |
| 03:08.07 | p3nguin | It doesn't specifically say 3-way calling is supported. |
| 03:11.01 | sweetpi | I didnt expect anything to do it automatically, I figured I would have to set it up. Maybe I'm just being naive. I thought it should be fairly simple to make a 3-way call on the same line |
| 03:12.27 | sweetpi | maybe I should just make a minicom script :p |
| 03:12.38 | p3nguin | The hardware will have to support making the 3-way call. |
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| 03:20.23 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:23.14 | crab | can someone comment on using IAX2 to connect two distant asterisk servers, one of which has a dynamic IP? is it a bad idea to even try, as i've been told? |
| 03:25.54 | brookshire | crab: i don't think so... as long as the IAX2 server is static it should be fine |
| 03:28.57 | crab | brookshire: what happens if a softphone on the server side tried to call an extension on the remote trunk side when the client's ip has changed? |
| 03:30.00 | crab | i mean, my local asterisk connects to the remote asterisk, but when someone tries to call me, my connection drops and my ip has changed. what happens before i re-establish the iax2 connection? some sort of "not reachable" signal? |
| 03:30.31 | ChannelZ | depends on your dialplan |
| 03:30.55 | ChannelZ | The actual device would be unreachable and show up as 'congestion' probably to the calling side |
| 03:32.46 | crab | and once the iax2 peer reconnects, i assume retrying the call will just work. |
| 03:33.25 | crab | thanks. i guess the people who are insisting that a static ip is required on both ends are on crack. |
| 03:33.43 | WIMPy | You could add openvpn to reduce the impact. |
| 03:33.55 | brookshire | or dynamics dns |
| 03:34.31 | WIMPy | What would that help? |
| 03:34.39 | ChannelZ | they're not on crack but you will have little service outages when IPs change depending on how often you re-register for instance |
| 03:34.42 | brookshire | map to the hostname not the ip |
| 03:34.52 | brookshire | ip changes, so does the dns entry ;) |
| 03:35.45 | brookshire | usually if your service is up, the ips don't change for a while.. unless the provider is just doing something funky |
| 03:36.05 | WIMPy | The DNS change will probably take a lot longer than a register. Especially if you trigger it when neccessary. |
| 03:36.20 | brookshire | depends on the timeout |
| 03:37.02 | WIMPy | I haven't tried on IAX however. But with SIP a asterisk -rx "sip reload" in your ip-up should do the trick. Probably wors with IAX as well. |
| 03:37.48 | WIMPy | Or (as I said) openvpn. That will even prevent active calls from clearing. You will just have a short(ish) drop out. |
| 03:38.37 | brookshire | and lots of overhead |
| 03:39.41 | WIMPy | You can use it without encryption. |
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| 04:04.35 | crab | thanks. |
| 04:07.23 | jql | crab? |
| 04:11.00 | crab | jql? |
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| 08:14.51 | TobSnyder | can yomeone tell me why an incoming DID is set to another DID? |
| 08:14.52 | TobSnyder | http://pastebin.com/frcH2NnV |
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| 08:53.23 | doolittlework | i think you are setting the cle ti the 4383 number |
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| 10:48.35 | [sr] | howdy people |
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| 11:04.27 | Xtratoz0rz | I wonder if anyone can help me? Im very new to using Asterisk and im trying to make an outbound call using IAX however is does not work. It says that it regected connect attempt with requested/capability 0xc/0xc incompatible with our capability 0xe703 |
| 11:05.08 | Xtratoz0rz | I have been searching the net for a fix for a problem and i think its to do with the codecs.. but cant figure out how to fix it |
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| 11:18.28 | odenkos_ntbk | Xtratoz0rz: never used IAX, but you could try changing your preference of codecs on your local peer settings if you think that it's cause is codec incompatibility.. |
| 11:21.40 | Xtratoz0rz | I tried adding disallow=all and allow=gsm to the iax.conf but it brings up the same error |
| 11:22.00 | Xtratoz0rz | i have tried a few diffrent codecs but allways the same problem |
| 11:22.09 | Gugge | did you try allow=all ? |
| 11:22.21 | Xtratoz0rz | yea i tried allow=all also |
| 11:23.40 | odenkos_lunch | Xtratoz0rz: then that's not a codec issue I think |
| 11:23.55 | Xtratoz0rz | ok |
| 11:24.23 | Xtratoz0rz | the exact error is |
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| 13:26.22 | tuxxie | How can I isolate a call in the cli when there are alot of calls bein processed? |
| 13:26.47 | tuxxie | s/bein/being |
| 13:29.54 | coredumb | Hi where can i find some complex IVR examples? |
| 13:30.23 | *** join/#asterisk Hounddog (~IceChat7@122.175.75.136) |
| 13:31.02 | Hounddog | hi, i installed asterisk and am trying to connect my softphone... now i was just checking with netstat and i do not see that asterisk is listening anywhere on port 5060 for the extensions |
| 13:31.37 | Hounddog | http://www.pastie.org/1137861 |
| 13:37.20 | drmessano | netstat -ln | grep 5060 |
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| 13:37.47 | Hounddog | forget it... me idiot forgot to take the proxy out of the softphone... |
| 13:39.24 | drmessano | Regardless, netstat isn't going to show the UDP ports with your query |
| 13:39.37 | drmessano | 5060 UDP is what you would be looking for.. |
| 13:40.27 | freckle | anyone been using skype for asterisk around on here? |
| 13:41.51 | Hounddog | ahh ok |
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| 13:42.02 | tuxxie | How can I isolate a call in the cli when there are alot of calls being processed? |
| 13:42.59 | freckle | tuxxie: depends on how you can identify the call, i set verbose to 0 and then do a show channels and scroll till i find it |
| 13:44.24 | tuxxie | freckle, I am looking for more of a call trace to see here the call dies |
| 13:45.05 | freckle | do you want to see the SIP trace of the CLI output? |
| 13:45.21 | freckle | /s/of/or |
| 13:45.53 | tuxxie | cli out put |
| 13:47.39 | tuxxie | well and the sip |
| 13:49.05 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 13:49.18 | freckle | the sip is what will tell you more IMHO... for that i use ngrep |
| 13:49.33 | freckle | something like.... |
| 13:49.56 | freckle | ngrep -tq -W byline 504@ port 5060 |
| 13:50.11 | freckle | where 504 is the extension you want to monitor |
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| 14:02.32 | drmessano | freckle: I've been using SFA |
| 14:03.11 | coppice | SFA == nothing |
| 14:03.17 | freckle | drmessano: what's that? |
| 14:03.34 | drmessano | ... |
| 14:03.37 | drmessano | Didn' |
| 14:03.39 | freckle | duh, Skyke For Asterisk |
| 14:03.42 | drmessano | Didn't you ask ... |
| 14:03.44 | drmessano | Yeah |
| 14:03.49 | freckle | yes, sorry |
| 14:04.24 | freckle | so how are you managing to dial Skype users which are alpha/numeric usernames? |
| 14:04.54 | drmessano | I setup numeric extensions to dial specific users |
| 14:05.05 | freckle | like shortcodes |
| 14:06.19 | freckle | i was thinking of a simple database with a numeric code for each skype user, then a click to dial setup... |
| 14:06.27 | drmessano | Not really.. No different than you handle SIP, IAX, etc now |
| 14:06.30 | freckle | needs to be easy for co-workers to use |
| 14:06.53 | drmessano | Is every extenion not just referencing a SIP/IAX2/ETC user? Same deal |
| 14:07.26 | freckle | i want a user on the office system to be able to dial myskypeid |
| 14:07.41 | drmessano | Yes, we established that |
| 14:08.18 | freckle | so as they can only dial numeric I thought of using agi to translate a number to a skype id |
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| 14:09.16 | Gugge | freckle: if its a small number of skype id's, why not just an extension for each, just like for each sip device? |
| 14:09.25 | drmessano | ^^^^ |
| 14:09.36 | freckle | it could run to 1000's of skype id's |
| 14:09.42 | Gugge | so can sip devices |
| 14:10.12 | freckle | but with a db and agi you dont need to reload the system all the time |
| 14:10.30 | Gugge | you dont need to reload the system to change the dialplan |
| 14:10.36 | Gugge | you only need to reload the dialplan |
| 14:10.43 | freckle | Gugge: exactly |
| 14:10.50 | drmessano | So rather than having 1001 dial BigRoger123, they're gonna dial 24476437123 ? |
| 14:11.32 | freckle | drmessano: nope they will dial a 4 digit shortcode or look the user up on the intranet and click to dial |
| 14:12.53 | drmessano | and exten => 1001,1,Dial(Skype/BigRoger123) is not acceptable? |
| 14:13.22 | freckle | wh have a massive dialplan when you can acheive the same thing with a AGI? |
| 14:14.13 | drmessano | Why waste resources with AGI when you can just put it all in the dialplan? |
| 14:14.45 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
| 14:14.50 | freckle | I have never had a resource problem with AGI |
| 14:15.12 | freckle | around 8 million calls a year and it just works great |
| 14:15.13 | Gugge | why said anything about resource problems ? |
| 14:15.17 | drmessano | You mean it doesn't use any resources? |
| 14:15.31 | freckle | what resource then? |
| 14:15.55 | Gugge | its gonna use more resources |
| 14:15.58 | Gugge | no doubt about it |
| 14:16.03 | Gugge | if its gonna be a problem, likely not |
| 14:16.26 | odenkos_ntbk | hey all, do you think it is usefull to read the SIP RFC? |
| 14:16.32 | Gugge | but why not just use odbc to lookup directly from the dialplan? |
| 14:16.42 | Gugge | and skipping the AGI stuff |
| 14:17.03 | drmessano | freckle: EXECUTION of ANY command uses some sort of resource in a system. AGI is going to cost you more than something executed from the loaded dialplan, no doubt |
| 14:17.08 | freckle | it's a trade-off between the server resource (minimal) compared to human resource to manager dialplan |
| 14:17.08 | Gugge | odenkos_ntbk: that really depends on what you need :) |
| 14:17.25 | odenkos_ntbk | Gugge: common knowledge :) |
| 14:17.28 | freckle | odbc = yuck |
| 14:17.38 | Gugge | odenkos_ntbk: then reading anything is usefull :) |
| 14:17.55 | Gugge | freckle: then use MYSQL() if its a mysql db, or whatever else you use :) |
| 14:17.59 | odenkos_ntbk | Gugge: I'm trying to make a SIP client almost from scratch.. |
| 14:18.07 | Gugge | odenkos_ntbk: then read the RFC :) |
| 14:18.28 | freckle | Gugge: I am, I was justing seeing if any alternative... but heard non better |
| 14:18.28 | odenkos_ntbk | Gugge: getting ready to :) |
| 14:19.01 | Gugge | freckle: you could dial the skype id directly like 999-skypeid, from the phones (my snom phones can dial letters) |
| 14:19.15 | Gugge | and make a 999. extension that dials the skypeid |
| 14:19.25 | freckle | Gugge: my users are not that savvy, or care about dialling like that |
| 14:19.26 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-232.mia.bellsouth.net) |
| 14:19.30 | drmessano | freckle: You said you were going to use AGI, not MYSQL() |
| 14:19.50 | freckle | the AGI script calls a MySQL query |
| 14:20.03 | Gugge | you can call the query directly from the dialplan with MYSQL() :) |
| 14:20.12 | drmessano | Rather than having Asterisk do it natively? OK |
| 14:20.23 | freckle | yes but I can do other stuff if I call an external script |
| 14:20.39 | Gugge | i havent needed to do anything i couldnt do in the dialplan yet :P |
| 14:20.44 | freckle | like deciding who is allowed to dial skype etc |
| 14:21.10 | Gugge | but sure, if you know the scripting language better than the dialplan language, go for it |
| 14:21.20 | *** join/#asterisk ariel_ (~chatzilla@93-62-188-210.ip23.fastwebnet.it) |
| 14:22.02 | freckle | well for instance deciding on the call time based on a tariff lookup and prepaid balance |
| 14:22.28 | Gugge | which is all stuff you can lookup from the dialplan too |
| 14:22.35 | drmessano | Wouldn't that just be multiple MYSQL() queries from the dialplan? |
| 14:22.50 | Gugge | or one query, with a couple of joins :) |
| 14:22.57 | drmessano | yeah |
| 14:23.12 | freckle | probably but I can develop a script tonnes faster... horses for courses I guess |
| 14:23.37 | Gugge | as long as you know its gonna require more resources, go for it |
| 14:23.49 | Gugge | most servers has way to much power anyway :) |
| 14:24.09 | freckle | Gugge: it has never been a problem before in 5 years of offering a commerical ITSP service. |
| 14:24.22 | drmessano | Nobody said anything about a "problem" |
| 14:24.44 | freckle | if it's not a concern why mention it? |
| 14:25.32 | drmessano | It's a problem when resources are exhausted.. the fact that it's going to cost MORE resources to execute the AGI than via Dialplan doesn't address whether or not it becomes the point of "problem".. that's pretty much up to how your system scales |
| 14:26.04 | freckle | in my case it is not a concern |
| 14:27.43 | drmessano | If someone tells you carrying an elephant in your truck is going to use more gas, telling them "Ive never run out of gas" is almost as useless as telling them what color underwear you're wearing |
| 14:27.51 | freckle | taking the SIP registration off asterisk saves the most resource |
| 14:29.03 | freckle | YMMV |
| 14:30.03 | bougyman | if you take all the phone traffic routing off asterisk it would use even less. |
| 14:31.02 | freckle | bougyman: I have Kamailio does that for me |
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| 14:43.14 | bougyman | i only toyed with kamailio a bit, in the end we decided to write all our routing and config in ruby, where we were comfortable and most competent. |
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| 14:44.07 | freckle | bougyman: if it works for you cool |
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| 15:02.38 | Ad-Hoc | hi |
| 15:06.15 | odenkos_ntbk | as I am reading the RFC I wonder.. if I unregister a sip peer, and set it to not register to *, and then I call something like exten@asteriskserver, then that call is treated by * as an anonymous call, if I get it correctly? |
| 15:06.35 | russellb | maybe. :-) |
| 15:06.51 | russellb | the registration doesn't really have anything to do with it |
| 15:07.26 | russellb | whether you're registered or not, each call will be authenticated if it matches a peer entry that has a password |
| 15:08.00 | odenkos_ntbk | russellb: yeah I read that the registration is only used for incoming calls.. |
| 15:09.06 | odenkos_ntbk | so if someone has a peer named exactly like one of asterisk's peers and calls my * server it will not be treated as anonymous? (if i set host=dynamic in the peer ofcourse) |
| 15:10.11 | russellb | depends how you configure it ... |
| 15:10.20 | russellb | if it's type=friend ... probably |
| 15:10.27 | russellb | type=peer, then no |
| 15:11.06 | [TK]D-Fender | [11:08]<odenkos_ntbk>russellb: yeah I read that the registration is only used for incoming calls.. <- With * yes. Some other servers use this as a way to firewall un-authed calls |
| 15:11.40 | [TK]D-Fender | odenkos_ntbk: as in if 1 places stays reg'd it will refuse other reg's within a timeout period, and calls, etc |
| 15:11.52 | russellb | [TK]D-Fender: you need threaded IRC |
| 15:11.57 | [TK]D-Fender | odenkos_ntbk: and every call is authed with * |
| 15:11.57 | russellb | :-) |
| 15:12.15 | odenkos_ntbk | ok, thanks russellb and [TK]D-Fender :) |
| 15:12.30 | odenkos_ntbk | I'll go and read on.. |
| 15:13.24 | [TK]D-Fender | russellb: Basket-weave :) |
| 15:14.24 | russellb | underwater? |
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| 15:28.08 | rue_house | so, compared to an 80Kbit isdn stream, why is there so much latency in 100mbit voip to cuase so many echo problems? |
| 15:28.35 | rue_house | cause none of the key'd digital 80kbit isdn systems have echo problems |
| 15:31.31 | rue_house | its not conversion time, the isdn systems have to do convrsion |
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| 15:34.38 | [TK]D-Fender | russellb: 80kbit says NOTHING about latency. that is a straight-line path to the digital interface. |
| 15:34.43 | [TK]D-Fender | rue_house: rather |
| 15:34.46 | russellb | WHAT |
| 15:35.11 | [TK]D-Fender | russellb: Already corrected, you may re-adjust your panties now :p |
| 15:35.16 | rue_house | yes, but you would think at 100mbits, you could atleast match the latency of a 80kbit system |
| 15:35.24 | [TK]D-Fender | rue_house: Local LAN? |
| 15:35.27 | rue_house | yes |
| 15:35.41 | [TK]D-Fender | rue_house: And what are you using as your measurement for "latency"? |
| 15:35.43 | rue_house | no routers |
| 15:35.52 | [TK]D-Fender | rue_house: And VoIP doesn't HAVE echo |
| 15:35.52 | rue_house | I'm basing this on echo |
| 15:36.00 | rue_house | yes, the conversion does |
| 15:36.14 | rue_house | but none of the keyd isdn systems have that problem |
| 15:36.14 | [TK]D-Fender | rue_house: Echo happens at the TDM>VoIP interface level |
| 15:36.21 | Greek-Boy | [TK]D-Fender: Still no ops after all these years? |
| 15:36.23 | Greek-Boy | :-( |
| 15:36.24 | rue_house | pots->voip |
| 15:36.25 | Greek-Boy | tsk tsk |
| 15:36.32 | [TK]D-Fender | rue_house: The "conversion" is within a device. |
| 15:36.48 | [TK]D-Fender | rue_house: There isn't a LAN between the two |
| 15:36.53 | [TK]D-Fender | rue_house: it is one box. |
| 15:37.09 | [TK]D-Fender | rue_house: And all EC is not made equal |
| 15:37.20 | [TK]D-Fender | rue_house: Nor are all conditions |
| 15:37.22 | rue_house | the isdn has NO echo canceling |
| 15:37.34 | rue_house | say like a nortel 616 |
| 15:37.48 | rue_house | pots->isdn no echo |
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| 15:38.05 | [TK]D-Fender | rue_house: Digitial systems do handset EC at the PBX level for those handsets. |
| 15:38.16 | rue_house | if I understand you right, your saying the echo is caused by a bad pots interface |
| 15:38.19 | [TK]D-Fender | rue_house: Built in |
| 15:38.32 | [TK]D-Fender | ~echo |
| 15:38.33 | infobot | somebody said echo was an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 15:38.57 | rue_house | there isn't echo canceling in those systems tho, its ulaw timeslice conversions |
| 15:39.06 | [TK]D-Fender | rue_house: Used to be a good link showing how impedence matching, etc plays into things. |
| 15:39.30 | [TK]D-Fender | rue_house: ANY 2-wire to 4-wire conversion needs this |
| 15:39.54 | rue_house | then shy dosn't a 616 have it |
| 15:40.11 | [TK]D-Fender | rue_house: Its built into the interfaces. |
| 15:40.18 | [TK]D-Fender | rue_house: it isn't an "option" |
| 15:40.20 | rue_house | its not |
| 15:40.24 | rue_house | its not there |
| 15:41.03 | rue_house | a 616 set and ksu dosn't have any echo canceling in it, it dosn't ahve the problem in the first place, by which if thats what you mean |
| 15:41.28 | rue_house | the isdn system has as much echo canceling as a pots set |
| 15:42.23 | rue_house | the high point being the speakerphone muting in speakerphone mode |
| 15:43.03 | rue_house | you can take two isdn system reciever and back to back them and they will squeel like a pig |
| 15:43.05 | [TK]D-Fender | ruethat is AEC, not far-side ECno |
| 15:43.30 | rue_house | yea, thats my point |
| 15:48.53 | rue_house | I wonder what the prop delay IS on an isdn system |
| 15:49.46 | rue_house | (when I say echo, I do mean the uses perception of it) |
| 15:50.44 | rue_house | what makes it all worse is that users start shouting into voip phones when they start sensing echo |
| 15:51.03 | rue_house | the isdn systems are all fixed gain |
| 15:51.39 | rue_house | which is a challange with voip, there is no way to even monitor the rtp levels on a voip call |
| 15:51.52 | *** join/#asterisk puzzled (~patrick@535335AA.cable.casema.nl) |
| 15:52.12 | rue_house | you have a audio line with gain controls on both ends, and a distortion box in the middle |
| 15:53.17 | rue_house | I think part of the key may be that voip systems do linear analog->digital conversion, and THEN convert it to ulaw |
| 15:53.29 | rue_house | isdn systems use ulaw adc/dac's |
| 15:54.36 | rue_house | but I cant prove that as I dont have info on a 'voip' pots card |
| 15:56.54 | rue_house | dont most voip sets have an adjustable packet timeslice setting? |
| 15:58.28 | rue_house | I think with an isdn system, your prolly only a few samples between encoded and decoded, and thats at 2Khz samples |
| 16:04.43 | rue_house | before I try to write a rtp stream level monitor, is there one out there? (there wasn't last I looked) |
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| 16:07.49 | [TK]D-Fender | [11:53]<rue_house>I think part of the key may be that voip systems do linear analog->digital conversion, and THEN convert it to ulaw <- ULAW = digital. |
| 16:08.11 | [TK]D-Fender | rue_house: Anything else is a codec conversion which has nothing to do with echo, etc. This IS the source |
| 16:08.45 | [TK]D-Fender | [11:54]<rue_house>but I cant prove that as I dont have info on a 'voip' pots card <- You have one already |
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| 16:35.39 | DelphiWorld | hi |
| 16:35.44 | DelphiWorld | i am calling through sip |
| 16:35.49 | DelphiWorld | from sip to asrterisk |
| 16:35.55 | DelphiWorld | and then go out through iax |
| 16:35.57 | DelphiWorld | but no audio |
| 16:36.02 | DelphiWorld | what could by the problem |
| 16:36.04 | DelphiWorld | , |
| 16:36.10 | DelphiWorld | doe asterisk iax2 need rtp ports? |
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| 16:37.16 | rue_house | [TK]D-Fender, there is a difference in linear adc and then recoding to ulaw as opposed to a ulaw adc |
| 16:39.18 | p3nguin | delphiworld: Check your NAT stuff on the SIP calls. |
| 16:39.33 | DelphiWorld | p3nguin: local, no nat |
| 16:42.05 | DelphiWorld | p3nguin: iax is remote but sip is local |
| 16:43.47 | p3nguin | IAX2 usually passes NAT without issue, so I wouldn't concentrate my energy on the IAX2 side of things. I would be trying to troubleshoot both sides independently. |
| 16:43.55 | DelphiWorld | p3nguin: evean betwan iax & sip users no audio |
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| 16:46.45 | ruben23 | hi guys |
| 16:51.04 | DelphiWorld | any help? |
| 16:55.01 | p3nguin | I gave you something to do, but you haven't completed it yet. |
| 16:56.55 | DelphiWorld | p3nguin: what something? repit it? |
| 16:57.50 | p3nguin | "I would be trying to troubleshoot both sides independently." |
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| 18:11.48 | jamko | http://pastebin.com/eSfEc7ZQ ..... this an example of a sip.conf that I am trying to get into realtime static. I am stumped on the correct values for "category" in the authentication section of sip.conf... Does each UA entry become it's own "category" or does every UA fall under the "Authentication" category? |
| 18:11.58 | *** part/#asterisk voip_troll (~les@96.51.239.24) |
| 18:12.26 | p3nguin | Maybe its own category. |
| 18:12.58 | jamko | lines 85-88 are the most relevant to my question. |
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| 18:19.09 | jamko | Yea most likely... which would mean authentication would be a one time category with no var_name or var_val .. I guess I will just have to try it. |
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| 18:41.49 | Hounddog | hi, i have a problem with my asterisk... i registered an extension but whenever i trie to call outside i just get this error.... WARNING[1493]: chan_sip.c:12334 handle_response_invite: Received response: "Forbidden" from '"station 100" <sip:0000000000@192.168.1.5>;tag=as7563cc8e' |
| 18:42.08 | Hounddog | i currently have no idea how and where this can happen |
| 18:43.35 | Hounddog | if someone could help on this i would really appreciate it |
| 18:44.05 | Hounddog | this is running on a ubuntu server scratch installation with asterisk 1.4 |
| 18:48.34 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
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| 18:58.57 | ChannelZ | well we need to see the sip peer from sip.conf first of all |
| 18:59.19 | Hounddog | ChannelZ sure :) |
| 18:59.31 | Hounddog | http://www.pastie.org/1138289 |
| 18:59.40 | Hounddog | this is the sip.conf |
| 19:02.24 | ChannelZ | and what are you dialing, which peer is it - would help to see the complete verbose console output |
| 19:02.42 | Hounddog | you want the sip debug? |
| 19:03.58 | ChannelZ | maybe but just the console with some verbose turned on (3 or 4) is a start. We have no idea what you dialed, what you dialed it from, etc. |
| 19:04.23 | Hounddog | ok |
| 19:04.27 | ChannelZ | I'm assuming at least it came from '400' since it's the only thing in your config |
| 19:04.38 | Hounddog | verbose 3or 4 would be the asterisk -vvvvvvg ? |
| 19:05.11 | ChannelZ | yeah or 'core set verbose 4' |
| 19:05.40 | ChannelZ | btw remove the quotes from your callerid string in sip.conf |
| 19:05.40 | Hounddog | ahh ok :) |
| 19:05.42 | Hounddog | one sec |
| 19:08.01 | Hounddog | http://pastie.org/1138337 |
| 19:08.57 | ChannelZ | who is SIP/testcarrier ? that wasn't in your config |
| 19:09.13 | Hounddog | oh sorry... there is another config |
| 19:09.16 | ChannelZ | it's coming from whatever that is though |
| 19:09.49 | *** join/#asterisk m0t3jl (~petr.mote@193.85.113.247) |
| 19:10.26 | Hounddog | http://pastie.org/1138344 |
| 19:10.40 | Hounddog | this is vicidial config... just realised that the carrier is there |
| 19:11.00 | Hounddog | i was thinking this is an asterisk problem maybee |
| 19:11.11 | Hounddog | i compiled everything from scratch... |
| 19:11.23 | ChannelZ | well RE: it's them sending back the 'forbidden' so you'd have to ask them why. |
| 19:11.41 | Hounddog | you mean vicidial? |
| 19:11.45 | ChannelZ | yes |
| 19:11.53 | Hounddog | hmpf |
| 19:12.08 | Hounddog | so it is no error or anything in asterisk you think |
| 19:13.09 | ChannelZ | well it could be a configuration error on your end, but the rejection is coming from them |
| 19:13.22 | Hounddog | hmmm |
| 19:13.53 | Hounddog | grmbl |
| 19:14.00 | Hounddog | this could be a tough one |
| 19:14.38 | Hounddog | i will just check something |
| 19:15.36 | ChannelZ | next you can look at sip debug and see specifically what it's returning |
| 19:16.10 | Hounddog | will do that... |
| 19:16.17 | ChannelZ | it might not like your auth - they might be looking at your callerID number for the DID and not getting it |
| 19:16.22 | Hounddog | i am just copying my sip conf from one server to this one... |
| 19:16.34 | Hounddog | on the other server it is working |
| 19:17.27 | *** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net) |
| 19:18.42 | Hounddog | different error at least lol |
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| 20:17.18 | Hounddog | ChannelZ i have changed some things but now i get this... maybee you know about that... dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:22.27 | Dovid | Asterisk can't get to the SIP Device |
| 20:22.32 | Dovid | prob. not registered |
| 20:24.03 | Hounddog | hmmm |
| 20:24.14 | Hounddog | you mean to the xlite softphone |
| 20:24.31 | Hounddog | i'l just check if it is registered |
| 20:26.03 | Hounddog | can i see which phone is currently registered? |
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| 20:27.03 | Hounddog | Dovid in sip show subscriptions it is showing nothing |
| 20:27.13 | Hounddog | i guess that could be it or? |
| 20:28.33 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:29.42 | Hounddog | http://pastie.org/1138460 |
| 20:29.49 | Hounddog | Dovid is registered |
| 20:33.51 | [TK]D-Fender | voip_troll: Hounddog Where is the complete failed call + CLI dump for that device as well as "sip show peers" to match |
| 20:33.53 | [TK]D-Fender | ? |
| 20:34.35 | Hounddog | one sec |
| 20:35.09 | Hounddog | http://pastie.org/1138476 |
| 20:35.11 | Hounddog | sip show peers |
| 20:35.53 | Hounddog | http://www.pastie.org/1138478 |
| 20:35.59 | Hounddog | cli dump |
| 20:36.04 | Hounddog | getting the sip debug |
| 20:37.03 | Hounddog | [TK]D-Fender http://www.pastie.org/1138481 :) |
| 20:38.01 | Hounddog | i'l get also the configs if required |
| 20:39.22 | [TK]D-Fender | [Sep 5 02:06:14] -- Executing [0012023267300@default:2] Dial("SIP/cc100-08f2bd98", "SIP/abhas/0012023267300") in new stack |
| 20:39.24 | [TK]D-Fender | [Sep 5 02:06:14] WARNING[9825]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:39.31 | [TK]D-Fender | abhas is not reachable, this isn't your PHONE. |
| 20:39.49 | [TK]D-Fender | Hounddog: You showed up cc100. That isn't what gave the error |
| 20:40.15 | [TK]D-Fender | abhas/51037 (Unspecified) D N 0 UNKNOWN <----------------------- |
| 20:40.25 | [TK]D-Fender | Hounddog: No way to get to abhas |
| 20:40.35 | Hounddog | hmmm |
| 20:41.14 | [TK]D-Fender | [16:24]<Hounddog>you mean to the xlite softphone <- no he didn't He meant whatever you CALLED which was clearly not the phone PLACING the call |
| 20:41.34 | Hounddog | the phone placing the call is the cc100 extension |
| 20:41.38 | Hounddog | 1 sec |
| 20:41.58 | [TK]D-Fender | Hounddog: I know that. THAT is fine. The peer you are CALLING is not. |
| 20:42.13 | Hounddog | ahh ok |
| 20:42.50 | Hounddog | which means something in my config is screwed |
| 20:44.42 | [TK]D-Fender | Hounddog: Nothing tells me that. |
| 20:45.39 | Hounddog | hmmm |
| 20:45.43 | Hounddog | i am confused |
| 20:46.25 | *** join/#asterisk ahowlader (howlader@180.234.42.222) |
| 20:46.42 | [TK]D-Fender | Hounddog: you never suitably describe WHAT "abhas" is and how you SHOULD be interacting with. So why should I think your configs are wrong? |
| 20:47.05 | Hounddog | abhas is the carrier information |
| 20:47.24 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 20:48.03 | [TK]D-Fender | Hounddog: The only reason * wouldn't have an IP is because you didn't set a fixed HOST for it which is what you do normally. Providers are fixed |
| 20:48.18 | [TK]D-Fender | Hounddog: your PHONE are usually mobile (dynamic) |
| 20:49.15 | Hounddog | http://pastie.org/1138501 |
| 20:49.20 | Hounddog | this is abhas currently |
| 20:50.54 | [TK]D-Fender | Hounddog: You have 2 HOST LINE in there |
| 20:51.05 | [TK]D-Fender | Hounddog: You set it up top and screwed yourself a few lines lower |
| 20:51.31 | Hounddog | argh |
| 20:51.35 | Hounddog | i see what you mean |
| 20:51.42 | Hounddog | host dynamic... |
| 20:51.46 | Hounddog | one sec lol |
| 20:53.06 | Hounddog | [TK]D-Fender YOU SAVED MY DAY |
| 20:53.19 | Hounddog | [TK]D-Fender can i donate some money to you? honestly... |
| 20:53.36 | Hounddog | and btw... to make you feel sad... i was paying someone to set it up :( |
| 20:53.38 | [TK]D-Fender | Hounddog: Always welcome |
| 20:54.21 | *** join/#asterisk DelphiWorld (~Delphi@41.200.7.7) |
| 20:54.28 | DelphiWorld | p3nguin: :P |
| 20:54.36 | p3nguin | hmm? |
| 20:54.46 | DelphiWorld | p3nguin: i just back |
| 20:54.53 | p3nguin | Solved the issues? |
| 20:54.59 | *** join/#asterisk Math (~mrene@freeswitch/developer/Math) |
| 20:55.02 | *** part/#asterisk Math (~mrene@freeswitch/developer/Math) |
| 20:57.39 | DelphiWorld | p3nguin: still bothring myself |
| 20:57.46 | DelphiWorld | p3nguin: at least you know i am freeswitch fan |
| 20:57.58 | p3nguin | I did not know that before. |
| 20:58.57 | DelphiWorld | p3nguin: ;) |
| 21:14.21 | DelphiWorld | p3nguin: please could you pastebin original sip.conf & extension.conf? |
| 21:14.36 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 21:15.14 | p3nguin | I don't know what you mean. |
| 21:15.25 | DelphiWorld | p3nguin: original sip.conf (unmodified) |
| 21:15.32 | p3nguin | I don't know what that is. |
| 21:15.48 | p3nguin | Are you talking about the samples that come with asterisk? |
| 21:15.49 | DelphiWorld | p3nguin: 1.4* |
| 21:15.57 | DelphiWorld | p3nguin: yes |
| 21:16.03 | p3nguin | There are no "default" config files. |
| 21:16.21 | p3nguin | I can look for the samples for you, though. |
| 21:16.43 | DelphiWorld | yes, please, p3nguin;) |
| 21:17.25 | [TK]D-Fender | DelphiWorld: Why don't YOU go look at the samples? |
| 21:17.31 | [TK]D-Fender | DelphiWorld: Go download them |
| 21:17.44 | DelphiWorld | p3nguin: i didn't know it, URL please? |
| 21:18.25 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 21:18.51 | *** join/#asterisk schreiber1337 (3f75ee02@gateway/web/freenode/ip.63.117.238.2) |
| 21:20.24 | schreiber1337 | Can anyone help me understand where "myprovider-out" comes from in the following line? |
| 21:20.32 | schreiber1337 | exten => s,n,Set(PEERCHECK1=myprovider-out) ; SIP peer name as defined in sip.conf |
| 21:20.52 | p3nguin | That's the peer name, as defined in sip.conf. |
| 21:21.32 | schreiber1337 | So, if I should replace that with the extension number that I want to check? |
| 21:21.37 | [TK]D-Fender | schreiber1337: It doesn't come from anywhere, you are setting a variable to hold that name. |
| 21:21.48 | [TK]D-Fender | schreiber1337: that line doesn't CHECK anything |
| 21:22.08 | [TK]D-Fender | schreiber1337: You are setting a variable and we have no idea what you are doing with it after |
| 21:22.14 | p3nguin | Extension numbers have little to do with peer names. |
| 21:24.00 | schreiber1337 | If trying to create a macro to check the status of a SIP Peer prior to paging over it's speaker... having a hard time determining the best way to check a phones status... any suggestions. |
| 21:24.29 | p3nguin | over it is speaker, hmm... |
| 21:25.08 | p3nguin | delphiworld: pastebinit has failed me, so I can't paste the sample file. |
| 21:25.11 | jamko | Does realtime allow for asterisk to share registrations across multiple boxes? |
| 21:25.26 | DelphiWorld | lol p3nguin |
| 21:25.55 | p3nguin | [rob@asterisk ~]$ pastebinit -i /etc/asterisk/extensions.conf.default |
| 21:25.55 | p3nguin | http://pastebin.com |
| 21:26.01 | p3nguin | NO URL ! |
| 21:26.12 | [TK]D-Fender | schreiber1337: We has NO idea what you coded. That 1 line setting a varibale means absolutely nothing to us. |
| 21:26.19 | p3nguin | I never had this problem before. Pastebin.com must have changed their stuff again. |
| 21:26.33 | schreiber1337 | OK.. .let me pastbin my macro |
| 21:26.39 | [TK]D-Fender | schreiber1337: You have not shown your code, or pointed out where things went wrong |
| 21:32.07 | DelphiWorld | [TK]D-Fender: any URL? |
| 21:32.27 | [TK]D-Fender | DelphiWorld: www.asterisk.org <-- |
| 21:37.36 | schreiber1337 | [TK]D-Fender: http://www.spectrumcontrol.com/zero/pastbin/PageMacro.txt Does this look like I'm at least on the right track? |
| 21:39.03 | [TK]D-Fender | schreiber1337: Dependsing how you call that... yes I guess |
| 21:39.11 | [TK]D-Fender | heads off for a while |
| 21:39.49 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 21:40.41 | DelphiWorld | p3nguin: exten => 883.,1,Dial(IAX2/mixvoip/${EXTEN}) |
| 21:40.43 | DelphiWorld | this ext is good? |
| 21:41.15 | p3nguin | not quite. |
| 21:41.17 | *** join/#asterisk ccesario_ (~ccesario@187.75.139.188) |
| 21:41.38 | p3nguin | Make it _883. or lose the . |
| 21:41.46 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 21:43.31 | *** join/#asterisk SP33D (~frank.l@unaffiliated/sp33d) |
| 21:49.06 | DelphiWorld | how do i use echo app? |
| 21:49.14 | p3nguin | Echo() |
| 21:49.41 | fenrus | exten => 1234,1,Echo() |
| 21:49.43 | fenrus | \o/ |
| 21:49.43 | DelphiWorld | p3nguin: exten => _555.,2,Echo() |
| 21:49.47 | DelphiWorld | p3nguin: good? |
| 21:50.06 | p3nguin | It could work. |
| 21:50.17 | fenrus | isnt it good practice to use "n" instead of numeric ordering ? |
| 21:50.38 | p3nguin | yes, it is. |
| 21:51.33 | p3nguin | http://pastebin.com/jBwuYXSh |
| 21:51.35 | DelphiWorld | didn't work |
| 21:51.47 | fenrus | what happens ? |
| 21:51.56 | p3nguin | Of course it didn't work. You dialed 555 thinking it would match that pattern. |
| 21:52.05 | fenrus | giggles. |
| 21:52.14 | p3nguin | 555 does not match that. |
| 21:52.29 | p3nguin | 5550 does. 5551 does. Et cetera. |
| 21:52.32 | DelphiWorld | p3nguin: is faling back to my trunk |
| 21:53.00 | DelphiWorld | ah |
| 21:53.03 | DelphiWorld | i forgot answer |
| 21:53.21 | p3nguin | No, you forgot to call a number that matched. |
| 21:54.31 | DelphiWorld | lol, p3nguin! no application echo! |
| 21:54.42 | p3nguin | uh |
| 21:54.49 | p3nguin | You don't have Echo() ? |
| 21:55.37 | DelphiWorld | p3nguin: note, this is not IBM server, but asterisk in openwrt |
| 21:55.41 | DelphiWorld | this is mini asterisk |
| 21:55.50 | p3nguin | Maybe you left it out? |
| 21:56.06 | DelphiWorld | p3nguin: not me but the package |
| 21:58.02 | DelphiWorld | p3nguin: how do i dial through sip? |
| 21:58.22 | p3nguin | Dial(SIP/peer/extension) |
| 21:58.48 | DelphiWorld | p3nguin: no, sip urp |
| 21:58.50 | DelphiWorld | uri* |
| 21:59.22 | p3nguin | Dial(extension@peer) or Dial(extension@peer.peerdomain.tld) |
| 22:00.17 | schreiber1337 | Anyone have a method for checking to see if an extension is in use? |
| 22:00.32 | p3nguin | ExtenSpy() or hints |
| 22:01.36 | schreiber1337 | I tried Hints... doesnt' seem to exist in the latest * ... |
| 22:02.03 | p3nguin | I'd be pretty surprised if it doesn't. |
| 22:02.28 | p3nguin | That would be almost like leaving out sip in the latest * |
| 22:03.29 | schreiber1337 | Well maybe I mis spoke... this is what I get.... " There are no registered dialplan hints" |
| 22:04.14 | schreiber1337 | Do I need to define something...somewhere to use hints |
| 22:04.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.82) |
| 22:05.17 | DelphiWorld | how do i cleare my sip registration? |
| 22:06.56 | *** join/#asterisk Diffen2 (~diffen2@81-234-228-6-no32.tbcn.telia.com) |
| 22:09.03 | DelphiWorld | how do i turn up sip traces? |
| 22:13.09 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:22.49 | DelphiWorld | p3nguin: working right now |
| 22:25.00 | DelphiWorld | [TK]D-Fender: success! |
| 22:26.55 | p3nguin | schreiber1337: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions#StandardPriorities |
| 22:32.02 | DelphiWorld | how do i set dtmf type? |
| 22:32.09 | DelphiWorld | doe iax2 support rfc2833? |
| 22:32.43 | p3nguin | I don't think iax2 has more than one mode. |
| 22:32.59 | *** join/#asterisk b0gatyr (~b0gatyr@adsl-149-109-188.mia.bellsouth.net) |
| 22:33.35 | DelphiWorld | p3nguin: so what mode is pocible? |
| 22:33.52 | p3nguin | the only mode. It is not changeable. |
| 22:34.32 | DelphiWorld | could by sip only? |
| 22:34.55 | p3nguin | In sip, you can change the mode with dtmfmode. |
| 22:35.29 | schreiber1337 | p3nguin: Thank you so much! I've been screwing around with different methods for hours now trying to make this work... |
| 22:36.22 | DelphiWorld | p3nguin: but dtmf cross iax2 & sip not working |
| 22:43.42 | DelphiWorld | any idea? |
| 22:45.14 | p3nguin | When things don't appear obvious, I prefer not to guess. |
| 23:26.33 | DelphiWorld | p3nguin: i think my dtmf issus will get fixed up on phone reboot |
| 23:26.42 | DelphiWorld | p3nguin: my phone was using inband audio |
| 23:26.47 | DelphiWorld | switching it to rfc2833 |
| 23:28.06 | DelphiWorld | p3nguin: worked! |
| 23:29.49 | DelphiWorld | :o/o |
| 23:38.54 | DelphiWorld | how do i answer in early_media? |
| 23:41.01 | DelphiWorld | could someone recomand a voip provider that provide iax trunking |
| 23:47.22 | p3nguin | voip.ms |
| 23:49.41 | p3nguin | That's who I use, anyway. |
| 23:49.51 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 23:54.01 | p3nguin | Are you in the US wanting to make most of your calls within the US? |