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| 02:59.16 | kikohnl | ~itsplist-us |
| 02:59.17 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 03:23.40 | lachesis | can anyone suggest a sip provider with a good pay-per-minute rate to USA numbers and support for caller id passthrough? |
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| 03:28.28 | [TK]D-Fender | <PROTECTED> |
| 03:28.35 | [TK]D-Fender | or vitelity |
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| 06:02.42 | mjt | I'm trying to call from one softphone to another, and that another is not currently present (the app is not running). The dialplan is just a single Dial(SIP/${EXTEN}). The callee does not hear anything while the server is trying to contact to (apparently saved somewhere) IP address of the other peer. |
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| 06:03.36 | mjt | I found that i can use "r" option for Dial() to force a ring-tone, but at the same time it has been said that asterisk provides this tone by its own in most cases, only sometimes it is necessary to force it. |
| 06:04.02 | mjt | so i wonder why this is necessary in the simplest case? |
| 06:06.02 | mjt | besides, the `r' option to Dial() generates a Very Loud sound, much louder than the subsequent call will be, can it be made a bit quieter? |
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| 06:34.55 | bulkorok | hi |
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| 06:35.10 | robert83a1 | hi all, I'm trying to load chan_capi.so with my Trixbox (yeah...I'm having fun) , it says loader.c error loading module chan_capi.so undefined symbol : manager_event |
| 06:35.18 | robert83a1 | is there a way to enable this fix this somehow? |
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| 06:57.24 | mjt | damn. In all examples on 'net, they use Dial() without the `r' option, and it apparently works... Why it doesn't work for me? :) |
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| 06:58.36 | mjt | and where (and why!) asterisk stored the ip address of the phone with a given extension? In sip.conf it is said to be host=dynamic |
| 06:58.57 | mjt | why it tries to connect to the old address more than 10 hours since that phone is turned off? |
| 07:02.00 | dwell | mjt: have you tried peer qualify=yes? |
| 07:05.14 | mjt | aha. that sounds interesting |
| 07:08.26 | mjt | yes, qualify=yes cleared the saved status, asterisk does not try connect to the old IP anymore. It looks like something like that should be set by default since any network disconnect will create leftovers |
| 07:09.29 | mjt | thank you dwell! |
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| 07:14.08 | kapil1941991 | Dear all , Any one knows how to configure a2billing with asterisk ? |
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| 08:56.55 | Blashyrkh | i try to use call files in combination with the email2fax script |
| 08:57.12 | Blashyrkh | but my asterisk seems to ignore the call file in the /var/spool/asterisk/outgoing directory |
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| 08:57.24 | Blashyrkh | do i need to set the location somewhere in the asterisk conf |
| 08:57.50 | Blashyrkh | is this the astspooldir in the asterisk.conf? |
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| 09:12.16 | Blashyrkh | okay i fixed it, i needed to put it in the outgoing directory in the spool directory |
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| 09:37.28 | Blashyrkh | i am trying to use the SendFAx() Application but i always get res_fax_spandsp.c:540 spandsp_fax_new: Channel 'DAHDI/124-1' FAX session '5' failed to create timing source. |
| 09:41.29 | Blashyrkh | okay so i need to install a timing source |
| 09:50.41 | Blashyrkh | i think i have timing sources modules installed |
| 09:50.49 | Blashyrkh | how can i check this on a running asterisk server? |
| 09:54.08 | kaldemar | module show like timing |
| 09:54.10 | kaldemar | timing test |
| 10:05.26 | Blashyrkh | mhm 0 modules loaded |
| 10:05.30 | Blashyrkh | thats teh reason i guess :D |
| 10:05.50 | Blashyrkh | but i just looked they wehre selected in menuselect |
| 10:06.35 | Blashyrkh | do i need to load them or activate them somehow? |
| 10:07.28 | Blashyrkh | res_timing_dahdi.so |
| 10:07.28 | Blashyrkh | res_timing_pthread.so |
| 10:07.28 | Blashyrkh | res_timing_timerfd.so |
| 10:07.28 | Blashyrkh | these three are present in my modules directory |
| 10:12.24 | kaldemar | check your /etc/asterisk/modules.conf for noloads concerning those. |
| 10:13.48 | Blashyrkh | are all modules loaded by default |
| 10:13.49 | Blashyrkh | ? |
| 10:16.09 | Blashyrkh | okay autoload is set to no, but i inserted a load statement for the mentioned modules |
| 10:16.37 | Blashyrkh | when i do modules reload they still dont show upmhm, i must have some stupid mistake somewhere |
| 10:19.15 | Blashyrkh | do i need some additionale moduls for the timing to work? res _realtime or something like this |
| 10:20.27 | kaldemar | don't load all timing resource modules, just one. |
| 10:20.51 | kaldemar | also, reload reloads modules, it does not load any new ones. module load <module> |
| 10:23.13 | Blashyrkh | thank you a lot, i am still new |
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| 12:10.41 | Ibrahim22 | Hi, is there a way for me to run a named extension after user enters a queue? |
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| 12:15.33 | Ibrahim22 | Hi, is there a way for me to run a named extension after user enters a queue? |
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| 12:17.37 | [TK]D-Fender | Ibrahim22: For whom? |
| 12:17.44 | [TK]D-Fender | Ibrahim22: To do what? |
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| 12:19.31 | Ibrahim22 | i want to get user in the queue, then have it ring on the user end (using PlayTones(ring)) for x seconds, and then start the music on hold of the queue. So I thought, I would put someone in the queue, and in the named extension stop the music on hold, use PlayTones(ring), use Wait(x), then start music on hold. |
| 12:20.34 | [TK]D-Fender | Either do those BEFORE they go to the queue (and then get queue MoH), or make a specific MoH class that includes that ringing etc as audio |
| 12:22.25 | Ibrahim22 | Yeah, i thought so too, but if i do PlayTones(ring), Wait(x), THEN Queue(...), that would mean, the user wouldn't be added to the queue until Wait(x) was finished |
| 12:22.55 | [TK]D-Fender | Correct. |
| 12:23.13 | [TK]D-Fender | but you don';t get to execute arbitrary stuff while IN the queue |
| 12:23.22 | [TK]D-Fender | So either waste time before entry or make it part of the MoH |
| 12:24.03 | Ibrahim22 | Okay, got it! Thanks |
| 12:24.34 | [TK]D-Fender | <PROTECTED> |
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| 15:12.28 | mjt | how to set/return a code to be displayed on the calee handset about the (unsuccessful) call status? |
| 15:13.19 | [TK]D-Fender | Depends on the handset obviously |
| 15:13.28 | mjt | I tried to catch _X., to say the number they dialed is invalid. Before, the handset displayed "404 not found" or something like that, now it displays nothing |
| 15:17.51 | [TK]D-Fender | So the phone siplayed SIP reasons... you haven't shown us what it is actually doing now... |
| 15:18.55 | WIMPy | You can only have either a message or an announcement. |
| 15:19.15 | mjt | the phone displays "404 / not found" on the screen. Asterisk logs this: |
| 15:19.16 | mjt | Call from '151' (192.168.88.66:58536) to extension '200' rejected because extension not found in context 'office-phone' |
| 15:20.10 | mjt | i added a rule to try to playback something of the same theme: exten => _X.,1,Answer(500),playback(invalid),hangup() |
| 15:20.43 | mjt | now the text message is gone but the helpful audio message is here. Can't it do _both_ ? |
| 15:21.25 | WIMPy | Not with SIP and not with Asterisk. |
| 15:22.01 | mjt | heh. interesting. Thats |
| 15:22.12 | mjt | <PROTECTED> |
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| 15:23.04 | [TK]D-Fender | [11:20]mjti added a rule to try to playback something of the same theme: exten => _X.,1,Answer(500),playback(invalid),hangup() <- you also can't do all this on one line either |
| 15:23.29 | mjt | sure, i just didn't want to use a pastebin when all the info can be put in one line on irc |
| 15:23.39 | mjt | to save people an extra step |
| 15:24.01 | WIMPy | And you might want to use the I extension instead if that cathall. |
| 15:24.36 | WIMPy | Err, that's i not I. |
| 15:26.34 | mjt | doesn't appear to work. I actually tried it before, and later found this: http://www.voip-info.org/wiki/view/Asterisk+i+extension |
| 15:27.14 | mjt | "The 'i' extension only gets fired when there's a prompt or input been made with 'background'" |
| 15:28.02 | WIMPy | Hmm. I would have sworn I did it that way. |
| 15:28.42 | mjt | actually that same page shows another possible solution down the road |
| 15:28.53 | WIMPy | Yes, I definitely do. |
| 15:29.19 | [sr] | hi |
| 15:29.34 | [sr] | hi, who has a linksys 3102 ? cant get inbound working |
| 15:29.36 | WIMPy | But it does seem to make a difference how and from where I dial an invalid number. |
| 15:30.05 | [TK]D-Fender | [11:27]mjt"The 'i' extension only gets fired when there's a prompt or input been made with 'background'" <- no. |
| 15:30.35 | [TK]D-Fender | mjHowever i is NOT used for SIP matching on an incoming call. It 404's that is all. You either answer... or you don't |
| 15:31.23 | mjt | the second variant in that voip-info page actually works |
| 15:31.46 | mjt | but indeed, it does not return the status code back still, only the audio message |
| 15:33.19 | c|oneman | ok, so I figured out that switchvox pushes older bootroms to my polycom phones. I'd like to manually provision phones using the latest bootrom (4.0.6), and then try to get the other provisionning settings as close as possible. How can I figure out which settings get pushed from switchvox? |
| 15:35.01 | WIMPy | mjt: First of all, you need to Hangup(1), but even if you try Playback with ,noanswer it's rather questionable if you get a message on you phone, but definitely *after* the announcement. |
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| 15:55.18 | mjt | heh |
| 15:56.01 | mjt | i was trying that here. indeed, at least some phones does not output any voice made by playback(,noanswer) |
| 15:57.32 | mjt | and yes i found Hangup(code) too. If I use hangup(1) after answer it got lost; if i use playback(,noanswer), hangyp(code) works but not the audio. I was also reading the SIP standard a bit, and indeed [TK]D-Fender is right, it is EITHER answer OR code. |
| 15:58.10 | mjt | is it not a frequently asked question? |
| 16:00.38 | mjt | i tried several android softphones, and many of them does not display those codes at all |
| 16:01.11 | mjt | so the default/sample asterisk config, where the codes are used, does not look very nice with them. |
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| 16:21.51 | mjt | IAX registration refused, cause code: 29. What can it be _except_ wrong username:password? |
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| 16:28.01 | [TK]D-Fender | mjt: Show us your peers masking only the secret |
| 16:30.21 | mjt | http://fpaste.org/94412/ -- that's complete with the secrets |
| 16:31.39 | mjt | for panda aka tlspz it is the whole iax.conf, for tls aka wh it is the sample config with one added register + one added section (it is rather large) |
| 16:35.59 | mjt | http://fpaste.org/94416/ -- complete iax.conf from both sides; on wh i removed comments by doing sed -e 's/;.*//' -e '/^[ ]*$/D' iax.conf |
| 16:40.08 | [TK]D-Fender | mjt: You have HOSTS specified for BOTH sides. |
| 16:40.17 | [TK]D-Fender | You cannot register when you have defined the host. |
| 16:40.41 | [TK]D-Fender | The whole point ofregistration is to tell the other side where you are. |
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| 16:43.24 | mjt | hmm. I must be missing something very simple, some basics.. ;) |
| 16:47.58 | anonymouz666 | where's your book? |
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| 17:15.37 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 17:24.43 | mjt | ok. so the two nodes see each other. registration wents fine. iax2 show registry and show peers shows good info. Using this dataplan: "exten => 102,1,Dial(IAX2/tls/${EXTEN})", that node does: |
| 17:24.57 | mjt | Executing [102@office-phone:1] Dial("SIP/151-0000000f", "IAX2/tls/102") in new stack; Called IAX2/tls/102; Hungup 'IAX2/tls-17939' |
| 17:25.09 | mjt | debug on tls does not show anything |
| 17:25.53 | mjt | so it hangs up immediately, even without trying to reach the otheer side (where extension 102 is registered) |
| 17:28.41 | mjt | the phone displays 503 / service unavailable |
| 17:32.31 | [TK]D-Fender | [13:25]mjtso it hangs up immediately, even without trying to reach the otheer side (where extension 102 is registered) <- this has nothing to do with 102 |
| 17:32.56 | [TK]D-Fender | And you are not looking in any detail |
| 17:32.59 | *** join/#asterisk acassio (~acassio@177.20.3.3) |
| 17:33.00 | mjt | more likely with tlspz=>tls connection (or lack therof) |
| 17:33.17 | mjt | detail? |
| 17:33.58 | [TK]D-Fender | No proper debug from each side |
| 17:34.03 | [TK]D-Fender | no peer dumps |
| 17:34.08 | [TK]D-Fender | nothing for us to comment on |
| 17:37.05 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 17:41.23 | mjt | heh |
| 17:41.24 | mjt | Found peer. What's device state of tls? addr=-1062686457, defaddr=0 maxms=2000, lastms=16 |
| 17:41.29 | mjt | nice addr ;) |
| 17:47.44 | mjt | type=user vs type=peer in iax.conf |
| 17:49.55 | mjt | ow. so my new server is already being probed for extensions |
| 17:50.04 | Nugget | [file] type=muffin |
| 17:51.18 | mjt | and the scanning restarts from the beginning in 2..3 seconds after i restart asterisk |
| 17:51.53 | mjt | i know ssh is being probed for many years already, but not at that scale |
| 17:56.01 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 18:07.35 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 18:17.13 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 18:21.11 | *** join/#asterisk Manu18 (52f6caae@gateway/web/freenode/ip.82.246.202.174) |
| 18:21.17 | Manu18 | Bonsoir |
| 18:21.44 | Manu18 | J'ai fini toute ma configuration s'asterisk |
| 18:21.54 | Manu18 | la, je viens de mettre mes regles iptables |
| 18:22.04 | Manu18 | et depuis plus de son lors des appels |
| 18:22.11 | Manu18 | j'ai bien ouvert le 5060 |
| 18:22.16 | *** join/#asterisk jemidy (~jemidy@nat/digium/x-lgnkuicjjdduugoa) |
| 18:22.19 | Manu18 | de 10000 à 20000 |
| 18:22.29 | Manu18 | je ne comprends pas trop |
| 18:23.51 | [TK]D-Fender | TOUS udp <- |
| 18:25.40 | Manu18 | oui |
| 18:26.34 | Manu18 | http://pastebin.com/L98qd3bt |
| 18:27.28 | [TK]D-Fender | iptables -t filter -A OUTPUT -p udp --dport 10001:35000 -j ACCEPT <--- NON. |
| 18:27.37 | [TK]D-Fender | T'es pas en controle de LEUR port |
| 18:28.03 | [TK]D-Fender | iptables -t filter -A OUTPUT -p udp --dport 5060:5061 -j ACCEPT <-- mieux d'enlever ca aussi |
| 18:28.22 | Manu18 | d'accord |
| 18:28.28 | Manu18 | et je dois ajouter quoi à la place? |
| 18:28.40 | [TK]D-Fender | rien |
| 18:28.47 | [TK]D-Fender | Tu fais trop |
| 18:29.25 | Manu18 | je n'ai toujours pas de son par contre |
| 18:30.08 | [TK]D-Fender | Et on ne voit toujours pas votre configuration.... |
| 18:30.21 | Manu18 | je paste mon sip.conf |
| 18:30.33 | [TK]D-Fender | Et il nous manques tous les details de votre environment |
| 18:30.57 | [TK]D-Fender | copie tout sauf les "secret" |
| 18:31.07 | Manu18 | oui |
| 18:32.40 | Manu18 | http://pastebin.com/yVQeaunM |
| 18:34.58 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 18:37.34 | Manu18 | Il y aurait quelque chose de mal configuré [TK]D-Fender ? |
| 18:45.42 | Manu18 | D'apres toi ce serait un soucis de config ou d'iptables [TK]D-Fender |
| 18:48.24 | *** join/#asterisk imcdona (~Thunderbi@75-151-101-101-Washington.hfc.comcastbusiness.net) |
| 18:49.10 | [TK]D-Fender | Manu18: Est ce-que votre serveur est derriere un router NAT? |
| 18:49.27 | Manu18 | il est sur serveur dedié |
| 18:50.31 | [TK]D-Fender | Manu18: freephonie devrait etre nat=no |
| 18:50.38 | Manu18 | ok |
| 18:50.41 | Manu18 | je teste |
| 18:50.55 | [TK]D-Fender | et tous : directmedia=no |
| 18:52.29 | Manu18 | ERROR[24781]: res_rtp_asterisk.c:2311 ast_rtcp_write_rr: RTCP RR transmission error, rtcp halted: Operation not permitted |
| 18:54.38 | Manu18 | <PROTECTED> |
| 18:54.47 | [TK]D-Fender | pastebi l'appel COMPLETE. Les linges simple de meme sont inutile |
| 18:54.48 | Manu18 | Beaucoup d'erreurs d'un coup |
| 18:54.55 | Manu18 | ok |
| 18:55.33 | *** join/#asterisk morphic (~vagrant@177.16.206.30) |
| 18:57.02 | morphic | hi, I have 4 queues configured in my queues.conf, sometimes ppl call them and stay in waiting because noone answer, I can check it by fastagi in my python app and hangup or add the channel as a member on the queue if the case |
| 18:57.17 | morphic | it there some way to call asterisk applications with AGI or fastagi |
| 18:59.19 | Manu18 | http://pastebin.com/0FkH5EKx |
| 19:03.50 | morphic | http://www.voip-info.org/wiki/view/exec |
| 19:03.57 | morphic | thanks very usefull irc channel |
| 19:05.11 | Manu18 | tu y comprends quelque chose [TK]D-Fender ? car moi, pas grand chose |
| 19:12.42 | [TK]D-Fender | Retransmitting #4 (no NAT) to 192.168.1.45:5060: |
| 19:12.47 | [TK]D-Fender | Contact: <sip:asterisk@62.210.195.144:5060> |
| 19:13.16 | [TK]D-Fender | pas bon |
| 19:13.45 | [TK]D-Fender | il essayer do contacter un address LOCAL en disent le IP public comme adresse de retour |
| 19:15.25 | Manu18 | Donc quelque chose a configurer dans iptables? |
| 19:15.51 | [TK]D-Fender | non |
| 19:16.05 | [TK]D-Fender | vos peers sont mal identifier |
| 19:16.34 | Manu18 | Est ce que tu pourrais m'aider? |
| 19:17.17 | Manu18 | pourquoi ans iptable ca fonctionne? |
| 19:18.30 | [TK]D-Fender | arret avec iptables. |
| 19:18.41 | [TK]D-Fender | ton ASTERISK n'est pas bien configurer |
| 19:18.50 | Manu18 | ok |
| 19:18.53 | [TK]D-Fender | FLUSH ton iptables et regle ton configuration. |
| 19:18.58 | Manu18 | peux tu m'aider? |
| 19:19.19 | [TK]D-Fender | Veuiller lire l'echantillon qui vient avec Asterisk. |
| 19:19.25 | [TK]D-Fender | BBL |
| 19:26.12 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
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| 19:44.40 | roxlu | hey guys, can someone explain me the meaning of the c= lines in SDPs. I read the RFCs, but when we have a=candidate:* lines I don't see why we have a c= line too? |
| 19:46.31 | roxlu | (when used with ICE) |
| 19:59.17 | *** join/#asterisk imcdona (~Thunderbi@75-151-101-101-Washington.hfc.comcastbusiness.net) |
| 20:02.54 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.34) |
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| 20:07.15 | *** join/#asterisk spicyramen_ (~Adium@sjspeed.wiline.com) |
| 20:07.33 | spicyramen_ | anyone uses SIP CLF for * ? |
| 20:18.02 | *** join/#asterisk imcdona (~Thunderbi@75-151-101-101-Washington.hfc.comcastbusiness.net) |
| 20:21.58 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
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| 22:37.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:16.12 | *** part/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
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| 23:24.38 | *** join/#asterisk bitvilag (~HUNbitvi@62-165-202-131.pool.digikabel.hu) |
| 23:24.45 | bitvilag | hey everyone |
| 23:28.30 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 23:32.22 | bitvilag | just wondering how unsafe is to have asterisk sip port out in the open and rely on the auth of the extensions? |
| 23:33.37 | [TK]D-Fender | As opposed to? |
| 23:35.46 | *** join/#asterisk imox (~imox@p4FC5C66C.dip0.t-ipconnect.de) |
| 23:45.22 | bitvilag | well |
| 23:45.55 | bitvilag | my concern is if they manage to hack it easily my bills for the phoneline jumps up a bit:P |
| 23:46.00 | bitvilag | so how easy is it? |
| 23:50.56 | [TK]D-Fender | First they have to have a peer they can auth against, that means picking a valid account name. |
| 23:51.09 | [TK]D-Fender | Certain options make that very hard to confirm. Then there is the pass |
| 23:51.42 | [TK]D-Fender | A better security model involves look at this behavior and deciding to prevent it. |
| 23:51.50 | [TK]D-Fender | Most people use things like fail2ban for this |