| 00:10.03 | *** join/#asterisk kayfox (~kayfox@xheotris.zerda.net) |
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| 00:53.11 | p3nguin | wonderworld: It's the same idea as macro. Dial uses M for macro, and U for gosub. |
| 00:53.34 | p3nguin | wonderworld: core show application Dial |
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| 01:32.28 | *** join/#asterisk xpot-mobile (~xpot@166-70-100-198.ip.xmission.com) |
| 01:33.24 | xpot-mobile | question: is there such a file privacy-solicitors? some along the lines of "The party you are trying to reach does not accept solicitations"? |
| 01:33.47 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 01:43.47 | p3nguin | xpot-mobile: yes |
| 01:44.41 | p3nguin | extra-sounds-en.txt:privacy-if-sales-call-contact-in-writing and extra-sounds-en.txt:privacy-screening-unidentified-calls are relevant |
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| 01:50.26 | xpot-mobile | p3nguin: yes, but they do not specifically mention solicited calls... just unidentified calls. I want this specifically if possible "You have reached a number that does not accept solicitations. If you are a solicitor, please add this number to your do-not-call list and hang up now. Otherwise, please press 1, or stay on the line." |
| 01:51.19 | xpot-mobile | its a popular telecom audio file, surprised its so hard to find. |
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| 01:53.33 | wonderworld | maybe tt-monkeys would be the correct file to playback? |
| 01:56.13 | xpot-mobile | lol |
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| 02:19.54 | p3nguin | xpot-mobile: /var/lib/asterisk/sounds/en/extra-sounds-en.txt:privacy-screening-unidentified-calls: Due to the huge number of unsolicited calls we recieve, we are now screening all inbound calls where a caller ID is unavailable or withheld. |
| 02:20.11 | p3nguin | I guess that's not what you need. |
| 02:20.48 | xpot-mobile | no, that doesn't tell the caller to add the called number to their do-not-call list |
| 02:21.09 | p3nguin | Make your own. |
| 02:21.22 | xpot-mobile | going to, just wanted to see if one was already created ;) |
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| 03:49.49 | mscdex | anyone here familiar with getting the cisco 7961 to work with asterisk 1.8.x? i have the extension's 'nat' setting set to 'no', which sends the SIP responses back to 5060, but it seems the phone still does not see that message |
| 03:50.09 | mscdex | s/to/on/ |
| 03:50.56 | mscdex | er |
| 03:51.26 | mscdex | i should also note that a softphone (sjphone to be precise) on a desktop on the same network as the phone works flawlessly |
| 03:51.45 | p3nguin | Extensions have no concept of NAT. |
| 03:52.47 | mscdex | huh? |
| 03:54.25 | mscdex | there is a `nat` setting that is valid for each extension in sip.conf. changing its value to `yes`, `no`, or `never` does change the behavior of where the SIP response traffic goes |
| 03:55.01 | mscdex | without `no` it sends it back on the source udp port (a high udp port the cisco phone chooses) instead of 5060 |
| 03:55.34 | mscdex | and as far as i know the phone only listens on 5060 for responses |
| 03:56.10 | mscdex | but for some reason it's just not getting the response, despite the response having a destination udp port of 5060 |
| 03:56.18 | p3nguin | extensions are not in sip.conf; extensions are in extensions.conf. |
| 03:57.06 | mscdex | er well, i'm using freepbx and it is actually sip_additional.conf that it uses |
| 03:58.22 | mscdex | the one that defines the numerical extensions with the user info (secret, etc) |
| 03:59.00 | mscdex | i've been at this for days now and still cannot figure out what's wrong |
| 04:06.55 | irule | hi, I configured my spa3102 propperly last night, went to sleep, shut it off, and today IO cant access the thing again, it asks for password! no GUI :s |
| 04:09.04 | p3nguin | What, exactly, is asking for a password? |
| 04:10.34 | p3nguin | Like... your phone rings and a voice says, "Please enter your password," or what? |
| 04:10.38 | irule | well I recall that I created an admin password called "birdies" on the web admin, so I try 2473437 that corresponds to numeric birdies o the phone, it is incorrect :s |
| 04:10.57 | p3nguin | The web server isn't working? |
| 04:11.07 | irule | it asks for regular voice command, asks for password after I type to check a configuration |
| 04:11.34 | irule | no web server, not even the static IP I created, WAN 192.168.1.251, so no ping |
| 04:12.01 | p3nguin | Did you turn off the web server last time? |
| 04:12.23 | irule | I followed 3 different blog posts to make both sip channels register, it will be dreadfull to have to start all over again haha |
| 04:12.36 | irule | no I did not turn web server off |
| 04:12.40 | p3nguin | It's very simple to configure. |
| 04:13.23 | p3nguin | Is your LAN cable connected to the Ethernet or Internet port on the device? |
| 04:13.47 | irule | internet cable to the lan |
| 04:13.59 | p3nguin | That isn't what I asked. |
| 04:14.02 | irule | the blue one, no cable to the yellow ethernet |
| 04:14.03 | p3nguin | Your LAN... |
| 04:14.14 | p3nguin | You have a switch with a cable connected to it... |
| 04:14.30 | p3nguin | Does that cable connect to the Internet or Ethernet port on the device? |
| 04:14.37 | irule | yes, 16 port switch, would that be the trouble? |
| 04:14.40 | p3nguin | no |
| 04:14.45 | p3nguin | Just stay with me. |
| 04:14.51 | [TK]D-Fender | Who's on first? |
| 04:15.18 | p3nguin | I'm only going to try one more time and then I'm off. |
| 04:15.29 | irule | [spa3102] <---> [16 port switch] <---> [adsl modem] |
| 04:15.42 | p3nguin | Did you connect your LAN cable into the Internet or the Ethernet port of the device? |
| 04:16.12 | irule | oh I did connect my LAN cable into the blue internet port :) |
| 04:16.18 | p3nguin | Okay. |
| 04:16.57 | p3nguin | And you configured the WAN settings to a static address as mentioned above. |
| 04:17.18 | irule | yes I did static ip 192.168.1.251 |
| 04:17.27 | irule | and it is invisible today :s |
| 04:17.47 | [TK]D-Fender | irule: By default the web admin is BLOCKED on the "Internet" port |
| 04:17.49 | p3nguin | If possible, connect a computer to the Ethernet port of the device and see if it gives you an IP address on the computer. |
| 04:18.07 | irule | <PROTECTED> |
| 04:18.42 | irule | let me see... |
| 04:18.43 | p3nguin | If it gives you an address when using the device as a router, browse to 192.168.0.1 and see if you can get the web interface. |
| 04:19.23 | p3nguin | This assumes a default setting on the LAN with the DHCP server enabled (also default). |
| 04:21.50 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 04:23.35 | irule | it is not giving address to a windows xp connected to the yellow ethernet port |
| 04:23.57 | irule | I did reboot xp twice trying this |
| 04:24.21 | p3nguin | Set the IP address of Windows static to something like 192.168.0.2 and then try to browse 192.168.0.1. |
| 04:25.11 | p3nguin | It is possible that your web server is now disabled, although I have no idea how it could magically be disabled when you were using it one day ago. |
| 04:26.04 | p3nguin | If it is disabled, the only way I know to enable it is by the IVR (which requires a password) or with provisioning (which may also be disabled). |
| 04:26.39 | p3nguin | I don't know of any other way to factory reset the device. |
| 04:28.37 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 04:29.01 | zeshoem | whats the easiest was to get an asterisk setup up without messing with any config files? |
| 04:29.14 | irule | elastix? |
| 04:29.19 | jpsharp | Pay someone to do it :) |
| 04:29.31 | irule | just dont pay me yet hahaha |
| 04:29.43 | zeshoem | I looked at trixbox, but it seems old |
| 04:30.10 | irule | p3nguin no web server, no ping on 192.168.0.1 from xp 192.168.0.2 |
| 04:30.25 | irule | what have we learned from my experience? |
| 04:30.41 | p3nguin | zeshoem: AsteriskNOW <----- correct answer |
| 04:32.28 | zeshoem | Thank you p3nguin , looks exactly like what I was searching for |
| 04:32.54 | p3nguin | If you can't manage config files, choose the FreePBX option on the boot up screen. |
| 04:33.16 | p3nguin | But remember, we do not support FreePBX here in #asterisk. |
| 04:34.29 | irule | may you please recommend me a simple guide to configure my spa3102? |
| 04:34.45 | p3nguin | If you can't get into it, there's no configuration to be done. |
| 04:34.57 | irule | I am resetting |
| 04:35.00 | p3nguin | What you need is a guide to factory reset the device. |
| 04:35.06 | [TK]D-Fender | No guide required. |
| 04:35.17 | [TK]D-Fender | Should have a nice little button |
| 04:35.33 | [TK]D-Fender | And I'm not hearing any diagnostics done to see what IP the device is claiming |
| 04:36.01 | p3nguin | Where is the button? I have a 3102 in my hand and I can't find any button on it. |
| 04:36.19 | [TK]D-Fender | Was pretty sure it had a recessed reset button on it.... |
| 04:36.25 | irule | ****,877778#,1 |
| 04:36.26 | [TK]D-Fender | Could be mistaken with another model... |
| 04:36.34 | [TK]D-Fender | Worst case is a phone-based reset |
| 04:37.11 | lanning | EMP? |
| 04:37.47 | p3nguin | The IVR reset would still require the password, wouldn't it? |
| 04:38.03 | irule | yes |
| 04:38.08 | irule | :s |
| 04:38.25 | irule | but how will I prevent this to hapen to me next time? |
| 04:38.42 | p3nguin | And the password you entered was invalid, so how are you going to reset it? |
| 04:39.47 | irule | I just did by IVR :D |
| 04:40.02 | p3nguin | How can you without the password? |
| 04:40.04 | irule | --Option Successful, please hang up to... |
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| 04:53.23 | irule | may I manage my own provisioning? I asume it is a configuration backup file :s |
| 04:53.34 | *** part/#asterisk jsjc (~Adium@138.Red-83-60-133.dynamicIP.rima-tde.net) |
| 05:01.38 | p3nguin | You can make your own provisioning files, yes. |
| 05:10.47 | irule | cool :D |
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| 06:11.49 | din3sh | morning all |
| 06:14.32 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:14.34 | schmidts | good morning |
| 06:15.34 | din3sh | am having a strange problem, realtime sip peers can dialout but not receive calls, CLI says no route to destination |
| 06:16.52 | din3sh | leifmadsen hinted to set ignoreexpery to yes |
| 06:16.52 | din3sh | which i did, problem still here |
| 06:16.52 | din3sh | :/ |
| 06:18.09 | schmidts | din3sh are these sip peers behind nat and do you have qualify=yes? |
| 06:19.05 | din3sh | no nat |
| 06:20.07 | din3sh | i dont have any value for qualify in sip_buddies |
| 06:20.12 | din3sh | should i set it to yes? |
| 06:20.14 | schmidts | and what do you see within a sip show peers? |
| 06:21.00 | din3sh | i have set rtcachefriends=yes , so i see the sip peers which have been cached |
| 06:21.11 | schmidts | qualify=yes activates a "pinging" method where asterisk sends every 60 seconds a sip OPTIONS message to the phone, to see if its still there, normally if the phones/peers are behind nat you use this to keep the port open, but you can also use it to test if the phone went unreachable |
| 06:21.45 | din3sh | ok i am in a non-natted environment |
| 06:22.09 | din3sh | what does qualify= defaults to if i dont set an explicit value? |
| 06:22.10 | schmidts | a no route to destination normally only happens if asterisk didnt know a way where to find a peer (ip + port) so this could normally only happens when a phone isnt either registered or unregistered from your server |
| 06:22.37 | din3sh | the phones can dialout though |
| 06:22.45 | din3sh | but not be called from other phones |
| 06:23.06 | din3sh | but the phone gets registered again by itself |
| 06:23.22 | din3sh | i can see on the CLI, registered.... |
| 06:23.42 | schmidts | and with sip show peer xyz (where xyz is your peer name) you see the right ip address of this client? |
| 06:25.21 | din3sh | no, says not found if the peer hasnt been cached (am using realtime) |
| 06:25.40 | din3sh | if its still cached, i see the IP address |
| 06:26.53 | din3sh | can i try to use qualify=yes even if am not on a nat |
| 06:35.36 | *** join/#asterisk nighty- (~nighty@209.119.68.98) |
| 06:39.16 | schmidts | sure |
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| 06:49.58 | haroldp | hiya |
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| 06:51.17 | haroldp | how can I reset a voicemail password? I edited voicemail.conf and restarted but it doesn't seem to be taking |
| 06:54.16 | kaldemar | haroldp: restarted what? a "voicemail reload" in CLI will do if you edit the right file correctly. |
| 06:54.59 | haroldp | restarted asterisk |
| 06:55.38 | haroldp | fixed it! thanks for your help! :) |
| 06:56.54 | haroldp | "rubber duck debug" heh |
| 07:00.22 | kaldemar | you don't need to restart whole asterisk, just reload the voicemail module. |
| 07:00.48 | haroldp | nods |
| 07:03.35 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 07:03.47 | madduck | any idea why pbx_config.so might be acting up? I cannot reload it, if it's loaded, a graceful restart kills asterisk with general protection fault, and asterisk ignores my extensions.conf |
| 07:03.54 | madduck | and the console has no 'dialplan reload' command. |
| 07:04.04 | madduck | version 1.6.2.9 |
| 07:05.10 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 07:05.39 | madduck | Unable to load module pbx_config.so |
| 07:05.43 | madduck | and on reload |
| 07:05.47 | madduck | The module 'pbx_config.so' was not properly initialized. Before reloading the module, you must run "module load pbx_config.so" and fix whatever is preventing the module from being initialized. |
| 07:05.51 | madduck | :*( |
| 07:08.40 | ChannelZ | That's bad. |
| 07:11.50 | *** join/#asterisk iulhk (~iulhk@119.152.40.63) |
| 07:11.57 | ChannelZ | Did you build asterisk yourself? |
| 07:12.35 | iulhk | having one linux machine centos, can i run multiple asterisk box at one machine ? |
| 07:13.44 | ChannelZ | short of virtualization, I guess you could maybe build multiple versions with different config paths, and you'd have to set them up on different port numbers... it'd be a mess |
| 07:14.25 | madduck | ChannelZ: no, Debian package |
| 07:14.29 | ChannelZ | actually I guess you could launch with -C and have different asterisk.conf's with different paths set there.. |
| 07:14.39 | madduck | ChannelZ: debug mode gives me |
| 07:14.40 | madduck | [Apr 3 09:13:42] DEBUG[6040] config.c: No file to parse: /etc/asterisk/extensions.conf |
| 07:14.44 | madduck | but the file exists! |
| 07:15.16 | ChannelZ | sounds busted. Permissions? |
| 07:16.32 | madduck | OMG |
| 07:16.41 | madduck | clobbers himself over the head with a GIANT BAT |
| 07:16.50 | madduck | ChannelZ: thanks a lot, it *was* as simple as permissions :( |
| 07:16.57 | ChannelZ | Oh good. |
| 07:17.25 | ChannelZ | better that than some crazy gremlin problem |
| 07:17.45 | madduck | thank you so much |
| 07:17.48 | madduck | you saved my day. |
| 07:17.58 | madduck | i was about to throw stuff at my monitor |
| 07:18.48 | ChannelZ | heh sure |
| 07:20.11 | din3sh | o.O |
| 07:20.35 | iulhk | <ChannelZ> short of virtualization, the line was for me ? |
| 07:20.51 | ChannelZ | yes |
| 07:21.24 | ChannelZ | Question, why do you want/need to run multiple instances on one machine? |
| 07:21.33 | iulhk | <ChannelZ>: no matter if i am already using virtual linux machine ? |
| 07:23.09 | *** part/#asterisk madduck (~madduck@debian/developer/madduck) |
| 07:24.05 | iulhk | <ChannelZ>: have multiple client everyone has some its on applications, every one want different database , so having different applications, so don't want to show other's user one application flow to other so that's y , at the start i want to start for testing purposes. |
| 07:26.45 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 07:26.59 | ChannelZ | Well you can run asterisk in a VM. If you want to run more than one inside a single instance (VM or not) then you'll need to go to great pains to separate configs... and you're going to have to run them on non-standard ports which may be a problem unless they are all internal type systems |
| 07:27.42 | haroldp | you could keep normal ports if they are on different ips |
| 07:28.12 | haroldp | seems like an ideal job for a virtual machine tho |
| 07:30.11 | ChannelZ | Yes to both |
| 07:33.04 | iulhk | ok |
| 07:33.28 | iulhk | means i should away from it now :) |
| 07:34.53 | din3sh | if you want to have sleepless nights, then i suppose more than one instance on a single server is the way to go :p |
| 07:36.34 | ChannelZ | sleep is overrated |
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| 07:38.06 | din3sh | but so needed :) |
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| 09:59.44 | mirelab | hello all, does anyone know or have a sintax for REALTIME_STORE(...)=.. function |
| 10:00.34 | mirelab | i can get records from RT engine with REALTIME_FIELD and HASH but can't seem to write anything at all :( |
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| 10:03.56 | din3sh | mirelab: what are you using it for? |
| 10:04.20 | mirelab | for storing into a mysql database with realtime driver :) |
| 10:05.43 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 10:07.08 | mirelab | din3sh: from CLI: CLI> realtime store dnis_list Dnis 999 DnisID 999 |
| 10:07.10 | mirelab | Stored RealTime record. |
| 10:07.25 | mirelab | but no record is stored in mysql database |
| 10:07.41 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:10.45 | mirelab | ok, Realtime works from CLI bit i can't seem to get right sintax from Dialplan or Ael code |
| 10:13.59 | *** join/#asterisk FreezeS (~kvirc@ip4-89-238-223-70.euroweb.ro) |
| 10:14.29 | FreezeS | hi guys |
| 10:14.42 | FreezeS | I'm trying to install a hfc-pci card in 1.4 |
| 10:14.52 | FreezeS | and installed dahdi-hfcs |
| 10:15.28 | FreezeS | the module loads and the cards are configured but now when I load chan_dahdi I get this: Unknown signalling method 'bri_cpe_ptmp' |
| 10:15.50 | FreezeS | any pointers on how to troubleshoot this ? |
| 10:15.54 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 10:16.13 | elliot98 | politely enters |
| 10:16.45 | elliot98 | what are issues involved when using a SIP phone over cellphone networks and smartphones? |
| 10:17.09 | FreezeS | elliot98: 3G has big latency |
| 10:17.30 | FreezeS | you must use big packetization |
| 10:17.53 | elliot98 | FreezeS: aha...would using a g729 help? |
| 10:18.27 | FreezeS | ah, of course |
| 10:18.28 | elliot98 | also, how is NAT involved in cellphone networks...do cellphone networks constantly change the IP of the phones? |
| 10:19.02 | WIMPy | If you have high latency, you should send smaller packets. |
| 10:19.12 | FreezeS | actually, the packetization is for jitter |
| 10:19.17 | FreezeS | 3G has a lot of it |
| 10:19.28 | WIMPy | FreezeS: Better use at least Asterisk 1.8. |
| 10:19.31 | FreezeS | at least the networks I have experience with it |
| 10:19.50 | WIMPy | Packetization if for dalay vs bandwidth. |
| 10:20.14 | FreezeS | WIMPy: I have a lot of customization on that asterisk and if I upgrade a lot of things will fail |
| 10:20.28 | elliot98 | standard packetization is around 20 milliseconds? |
| 10:20.34 | elliot98 | what about registering properly? |
| 10:20.49 | FreezeS | elliot98: what countries are you planning to deploy in ? |
| 10:22.13 | FreezeS | btw, about NAT, all networks I've worked with had it but I don't think it's the rule |
| 10:22.15 | elliot98 | I've been able to register the device, but wonder if the cellphone networks change the IP addresses, close ports, etc. so can lose the registration IP address |
| 10:22.42 | FreezeS | some countries/networks actively filter voip |
| 10:22.44 | Chainsaw | elliot98: Use STUN and cross your fingers on both hands. |
| 10:22.54 | FreezeS | or mess up the packets |
| 10:23.18 | FreezeS | **cough**arabic countries**cough** |
| 10:24.38 | elliot98 | how would setting up a VPN help? |
| 10:24.58 | Chainsaw | elliot98: It would encrypt the traffic, so that it can't be interfered with. |
| 10:25.10 | FreezeS | it would add latency however |
| 10:25.17 | Chainsaw | elliot98: Then again, SIP-over-TLS & SRTP might have the same effect. |
| 10:25.36 | elliot98 | and in terms of not having the servers chase after the IP address |
| 10:25.40 | *** join/#asterisk R4md4c (~R4md4c@unaffiliated/r4md4c) |
| 10:25.41 | FreezeS | the "best" way is to use TCP for the VPN |
| 10:26.05 | FreezeS | it decreases the voice quality but the traffic is very hard to identify as voip |
| 10:26.30 | FreezeS | if you have the vpn over UDP they might identify it by the chattiness |
| 10:26.43 | FreezeS | many small packets |
| 10:26.48 | elliot98 | the SIP server will have the same IP address of the registration throughout, regardless if the cellphone networks change the IP address |
| 10:27.22 | elliot98 | I mean, the SIP device's IP address won't change |
| 10:27.37 | R4md4c | I've downloaded AsteriskNOW live cd 1.7.1 and installed it but I cant access its webinterface using username/password freepbx/fpbx |
| 10:28.00 | R4md4c | What is the correct password to access asterisk webinterface |
| 10:28.01 | R4md4c | ? |
| 10:28.34 | elliot98 | but then again, the VPN would need to register properly if the cellphone network start changing the external ip address |
| 10:28.37 | din3sh | use elastix |
| 10:31.08 | R4md4c | any idea what is the correct default password for AsteriskNOW ? |
| 10:31.20 | FreezeS | elliot98: normally the VPN reconnects when the tunnel drops |
| 10:36.49 | din3sh | R4md4c: have u tried admin/admin? |
| 10:36.50 | elliot98 | FreezeS: whereas SIP registration depends on the SIP device involved, etc. |
| 10:37.20 | FreezeS | not really, normally a sip device registers continuously |
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| 10:37.55 | elliot98 | FreezeS: every couple of minutes or so |
| 10:38.08 | FreezeS | yes, depends on the configuration |
| 10:38.41 | elliot98 | how common is it though for cellphone networks to switch the external IP? |
| 10:40.15 | FreezeS | normally a network doesn't change your IP juse because |
| 10:40.29 | FreezeS | you get a new IP when you disconnect and reconnect |
| 10:40.42 | FreezeS | on mobile that can happen very often |
| 10:42.04 | elliot98 | FreezeS: and even if not reconnecting, but simply, change locations, or following some cellphone network routing policy |
| 10:42.09 | elliot98 | FreezeS: ? |
| 10:42.12 | FreezeS | no |
| 10:42.27 | FreezeS | they don't change your IP unless they have to |
| 10:42.45 | elliot98 | FreezeS: so even on a cellphone, unless reconnecting, the IP would stay the same. Is it usually NAT? |
| 10:42.56 | FreezeS | yes, usually it is |
| 10:44.18 | schmidts | elliot98 gsm handover works over gsm not over ip, so the ip connection is above the gsm connection which does the cell roaming |
| 10:45.39 | R4md4c | din3sh: Yes, It is not working |
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| 10:47.05 | elliot98 | wonder if cellphone networks close off ports faster than standard routers |
| 10:47.12 | din3sh | R4md4c: you able to ssh to the box though? |
| 10:47.58 | R4md4c | din3sh: Actually the problem is not logging to the system. the problem is in logging to asterisk's webinterface |
| 10:48.36 | din3sh | have u tried resetting the password? |
| 10:48.44 | din3sh | i mean is it possible via the ssh |
| 10:48.45 | din3sh | ? |
| 10:49.31 | R4md4c | din3sh: Yeah it is possible |
| 10:49.44 | R4md4c | I just sshed to the box |
| 10:54.01 | R4md4c | R4md4c: Do you know which conf file that has the usernames/passwords to the web interface ? |
| 10:54.09 | R4md4c | din3sh: ^^ |
| 10:57.00 | R4md4c | Okay found it. It was in manager.conf file and the username/password was admin/password |
| 10:57.24 | din3sh | etc/amportal.conf |
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| 11:08.00 | din3sh | R4md4c: ok? |
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| 11:09.03 | R4md4c | din3sh: I didnt find the /etc/amportal.conf file but I found /etc/asterisk/manager.conf which I found the default username/password which was admin/password |
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| 11:24.13 | din3sh | R4md4c: were u able to log in? |
| 11:24.38 | R4md4c | Yeah everything is okay for now :) thans alot |
| 11:25.38 | din3sh | np |
| 11:25.39 | din3sh | ;) |
| 11:28.30 | mirelab | does anyone know syntax for REALTIME_STORE function, i always get error:[Apr 3 13:27:35] ERROR[7453]: pbx.c:3524 ast_func_read: Function REALTIME_STORE cannot be read |
| 11:29.38 | Gugge | mirelab: show the dialplan |
| 11:30.54 | mirelab | Gugge:http://pastebin.com/Ye3dKYFV |
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| 11:31.14 | mirelab | Gugge: ael code |
| 11:31.20 | Gugge | dont use someting=REALTIME_STORE() |
| 11:31.27 | Gugge | use REALTIME_STORE()=something |
| 11:31.44 | mirelab | ok i'll switch |
| 11:31.53 | Gugge | REALTIME_STORE(family,field1,field2,...,field30) = value1,value2,...,value30; |
| 11:31.55 | Gugge | something like that |
| 11:33.33 | mirelab | Gugge: that is what i need, and i couldn't find syntax anywhere :) thanks |
| 11:34.02 | mirelab | Gugge: i tried with ARRAY, with () with "" and nothing worked |
| 11:35.01 | Gugge | :) |
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| 11:35.47 | mirelab | Gugge: finnaly thx a lot :) |
| 11:36.30 | mirelab | Guggenever would have tried only with (comma) seperated values |
| 11:36.40 | kaldemar | mirelab: the "Function REALTIME_STORE cannot be read" error comes from ${} around the function. use it with the Set application. |
| 11:36.45 | mirelab | without () or anything lol |
| 11:37.28 | mirelab | kaldemar: ok thx , I will, i need that 0 value if recodt is set |
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| 11:43.56 | leifmadsen | Set(Result=${FUNC_RETRIEVE(arg1,arg2)}) <-- way you get data from a function || way you set values to a function --> Set(FUNC_PUSH(arg1,arg2)=value1) |
| 11:43.59 | mirelab | http://pastebin.com/4hy8TsCT |
| 11:44.27 | mirelab | it's braking after first (comma) but record is there in DB |
| 11:44.27 | leifmadsen | mirelab: yes, you have a syntax error |
| 11:44.32 | leifmadsen | you don't have a closing ) |
| 11:45.38 | kaldemar | and no values |
| 11:45.44 | leifmadsen | that too |
| 11:45.52 | leifmadsen | you're just passing args, but have set no value |
| 11:45.56 | leifmadsen | see my syntax above |
| 11:46.15 | kaldemar | and no application. |
| 11:46.39 | mirelab | ok thx :) |
| 11:46.44 | leifmadsen | mirelab: read this --> http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-2.html |
| 11:47.33 | kaldemar | Set(REALTIME_STORE(family,field1)=value1) would be a simplified example on how to use the function. |
| 11:53.35 | mirelab | thx everyone for fast help :) |
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| 11:57.54 | Valken | Hi everybody. |
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| 11:58.41 | leifmadsen | ohai |
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| 12:27.54 | Valken | I need help with DTMF. I can not find any tutorial explaining how to route calls based on dtmf. |
| 12:28.15 | Valken | Does anyone of you can share a tutorial link? |
| 12:28.16 | Valken | Thank |
| 12:30.07 | kaldemar | try to explain what you want to do in other words. "route calls based on dtmf" is a bit vague. |
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| 12:36.14 | Valken | sorry, my english is not good at all. |
| 12:37.00 | Valken | typing the question, I think I got the answer |
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| 12:37.01 | Valken | brb |
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| 12:58.04 | [TK]D-Fender | kaldemar, For a sonata press 4, for a symphony press everything at once. Thank you! |
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| 13:04.59 | Valken | D-Fender: That was it, I found the proper word IVR. ;) |
| 13:05.31 | Valken | Is it wise to jump from context to context or is it better, at some stage to split the service between different asterisk server? |
| 13:06.12 | kaldemar | [TK]D-Fender: and for a cantata, yell! |
| 13:07.09 | kaldemar | Valken: depends on the service. |
| 13:07.34 | Valken | oki |
| 13:08.15 | Valken | I'm using the WaitExten entry but it wait too long after the dtmf signal has been sent. Any way to make it shorter? |
| 13:08.27 | Valken | I think there is a dtmf length but don't know where. |
| 13:10.51 | kaldemar | "core show function TIMEOUT" |
| 13:11.02 | WIMPy | If it waits, you have ambigous patterns in your dialplan. |
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| 13:12.31 | Valken | Hum: |
| 13:15.47 | Valken | Here it is: |
| 13:15.47 | Valken | http://pastebin.com/uFdknHzG |
| 13:16.13 | Valken | the sip account sends all calls to the context privatePhones |
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| 13:19.12 | [TK]D-Fender | Valken, Set the TIMEOUT's as kaldemar referred you to look at. |
| 13:21.27 | [TK]D-Fender | Valken, Also you are running your IVR off a PATTERNED extension that will interfere with the other "options" in that context. |
| 13:21.44 | [TK]D-Fender | Valken, Which is why people typically use "s" as the "base" for their IVR's |
| 13:21.47 | [TK]D-Fender | ~stdextens |
| 13:21.48 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 13:24.18 | Valken | Thank a lot for the help. I did use the s extension but ran into problem so I tried with the _X. |
| 13:24.25 | Valken | Back to the s extension now. :D |
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| 14:11.57 | bobb_WU | hello everybody |
| 14:12.14 | bobb_WU | i'm looking for a little help with defining realtime sip phones |
| 14:12.39 | bobb_WU | which columns are the most important to define? |
| 14:13.01 | leifmadsen | probably all of them |
| 14:13.12 | bobb_WU | definitely not all of them |
| 14:13.19 | bobb_WU | there's like 30 |
| 14:14.00 | bobb_WU | if its dynamic, asterisk will input into the ipaddr, yes? |
| 14:14.05 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html |
| 14:14.26 | leifmadsen | you were talking about defining the columns, not the values of the row |
| 14:14.40 | leifmadsen | use that link above |
| 14:16.45 | bobb_WU | thank you leif |
| 14:17.30 | bobb_WU | i am trying to define the values though. i have a few defaulted such as context and type |
| 14:17.47 | bobb_WU | from table 16.3 on that link, are those the important columns to have values? |
| 14:18.47 | leifmadsen | as it says, those are the minimum values to have defined |
| 14:19.29 | *** join/#asterisk jurij1 (~jurij@213.215.248.66) |
| 14:19.46 | jurij1 | i get No such command 'core show translation'. how can that happen? |
| 14:19.52 | bobb_WU | what is regserver? |
| 14:19.59 | jurij1 | i run asterisk 1.8 |
| 14:20.22 | jurij1 | please someone help me, google doesnt give any answer to this |
| 14:25.47 | kaldemar | "When a peer registers to the system, saves the systemname (as defined in asterisk.conf) into the regserver field within the database." |
| 14:26.03 | kaldemar | looks like the sentence is missing something but is still understandable. |
| 14:26.26 | leifmadsen | s/saves/Asterisk saves/ |
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| 14:40.40 | jurij1 | anyone? |
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| 15:03.06 | aberrios | jurij1, did you compile or install from a repo? |
| 15:03.43 | jurij1 | elastix 2.2.0 |
| 15:03.44 | jurij1 | :( |
| 15:04.28 | aberrios | jurij1, and you've asked elastix peeps? |
| 15:04.35 | jurij1 | dunno where |
| 15:04.58 | aberrios | http://www.elastix.org/index.php/en/component/kunena/ would seem like a good point to start |
| 15:05.32 | jurij1 | oh thanx |
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| 15:21.03 | fprior | Hi all; where can I change the "Interdigit timeout" ? |
| 15:23.20 | Faustov | in features.conf |
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| 15:29.17 | fprior | Faustov, is featuredigittimeout option ? |
| 15:29.47 | Faustov | correct, in ms |
| 15:31.25 | fprior | Faustov, is not only for featured codes ? |
| 15:33.28 | Faustov | not that I know |
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| 15:45.15 | [TK]D-Fender | fprior, "core show function TIMEOUT" |
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| 16:11.41 | brokennoze | I appear to have found a bug. If I pause a queue member on a single interface, any other calls to a different queue on the same interface don't get delievered to the handset! using 1.8.5.0 - has anyone else experienced this problem? |
| 16:12.44 | [TK]D-Fender | brokennoze, .... you are over 5 releases out of date |
| 16:12.58 | [TK]D-Fender | 1.8.11.0 <------ |
| 16:13.40 | *** join/#asterisk wonderworld (~ww@dsdf-4db54223.pool.mediaWays.net) |
| 16:14.24 | brokennoze | ok, I'll check the release notes for bug fixes. |
| 16:17.16 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 16:17.36 | Chainsaw | brokennoze: Seemingly unrelated fixes may have cleared up the behaviour. Upgrading is the only way. |
| 16:21.03 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:25.34 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:30.44 | leifmadsen | brokennoze: that's why a development system where you can reproduce the error and perform an upgrade to confirm or deny that something has already been resolved is the way to go |
| 16:31.04 | leifmadsen | you can file an issue against 1.8.5.0, but the first thing a bug marshal is going to ask is if you can reproduce on the latest release |
| 16:33.56 | pabelanger | leifmadsen: sounds like you are talking about change management |
| 16:34.07 | leifmadsen | pabelanger: omg maybe :) |
| 16:34.18 | leifmadsen | pabelanger: sounds like a certain project I've been watching lately |
| 16:34.45 | pabelanger | http://tinyurl.com/3mku6w5 |
| 16:35.32 | leifmadsen | heh |
| 16:38.30 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 16:38.44 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:42.13 | *** join/#asterisk celord (~celord@201.191.95.81) |
| 16:49.44 | drfreeze | Hey |
| 16:50.12 | *** join/#asterisk ermali82 (~ermalkris@213.207.42.45) |
| 16:50.22 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 16:50.58 | ermali82 | hi does UUI "useruserinfo" work on asterisk 10.3.0 ? |
| 16:51.35 | *** join/#asterisk brokennoze (~IceChat9@host81-137-172-233.in-addr.btopenworld.com) |
| 16:51.35 | ermali82 | i'm trying to get the value of ${USERUSERINFO} but it is empty |
| 16:53.22 | brokennoze | OK, I've updated to version 1.8.11- rc3. There is still a problem. If a pause a queue member in one queue, put another call into a different queue that contains the same queue member (i.e. Interface) the 2nd call is never presented. |
| 16:53.55 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 16:58.20 | *** part/#asterisk clintc (~clintc@n128-227-12-58.xlate.ufl.edu) |
| 17:00.12 | drfreeze | I am trying to setup a softphone at a clients |
| 17:00.51 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:01.28 | drfreeze | We have the phone registering just fine and it can initiate a call |
| 17:01.36 | drfreeze | But the call dies on pickup |
| 17:02.40 | brokennoze | drfreeze: firewalls for RTP? |
| 17:03.01 | *** join/#asterisk the_weard (~arthur@41-132-18-108.dsl.mweb.co.za) |
| 17:03.42 | drfreeze | So, looks like I don't even need to answer |
| 17:03.52 | drfreeze | and the phone hangs up after one ring |
| 17:04.05 | leifmadsen | look at the sip traffic and find out what is goin gon |
| 17:04.29 | the_weard | hi im struggeling, can aneyone here on asterisknow/freepbx how do i setup my mail server setting like my smtp server and wich alias mail it should mail from or to |
| 17:04.29 | drfreeze | brokennoze: The calls are internal and made over PRI |
| 17:04.32 | Valken | my last similar experience was due to a non free codec. (g727?) |
| 17:04.41 | drfreeze | the sip it on the local network only |
| 17:04.44 | the_weard | is it an universal set of settings maybe? |
| 17:04.59 | Valken | To make sure I'd disallow all codec and allow only a few (like gsm) on both side. |
| 17:06.02 | kaldemar | ermali82: it is not supported by default. you need to define SUPPORT_USERUSER in channels/sig_pri.c and recompile. |
| 17:07.22 | *** join/#asterisk orioni (~chatzilla@46.183.121.133) |
| 17:07.40 | orioni | hi to all , how to get the called number on a deadAgi ? |
| 17:08.00 | orioni | im doing kind of billing |
| 17:08.52 | ermali82 | kaldemar: it is defined and i see the output of dump_user_user() in HEX but cant put it in a variable ${USERUSERINFO} |
| 17:11.07 | drfreeze | Ok, when we Dial in to the Softphone, the softphone can pick up and the call remains active |
| 17:11.35 | Valken | drfreeze: have you tried an echo test? |
| 17:11.39 | drfreeze | But the caller can't hear the softphone and the caller voice is garbled |
| 17:11.51 | Valken | It could help you to first determine if this is one phone or the other. |
| 17:11.54 | drfreeze | very much like a Sylon on battlestar galactica. :) |
| 17:12.36 | drfreeze | Valken: do you have a suggested method for the echo test? |
| 17:12.51 | p3nguin | hmm |
| 17:12.54 | Valken | euh, yup... |
| 17:13.27 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:13.44 | Valken | in your dial plan, instead of calling the sip device. you can use the Echo() function |
| 17:14.21 | p3nguin | It's an application, not a function. |
| 17:14.32 | Valken | sorry for the misnaming |
| 17:14.42 | p3nguin | exten => 500,1,Echo() |
| 17:14.55 | p3nguin | Call extension 500, talk to yourself. |
| 17:15.08 | Valken | ah, extension is the called number. Sorry |
| 17:15.09 | drfreeze | ok, good idea |
| 17:15.12 | drfreeze | I'll try the echo |
| 17:15.14 | Valken | I forgot we are in telephony |
| 17:15.22 | Valken | :P |
| 17:15.39 | Valken | oups |
| 17:15.50 | Valken | what's the difference between an application and a function? |
| 17:16.16 | p3nguin | app_whatever.so vs. func_something.so ? |
| 17:16.25 | p3nguin | I don't understand the question, really. |
| 17:16.31 | Valken | oh sorry. |
| 17:17.08 | Valken | in fact, I haven't check how it was named. So, if the difference is just in the naming, you can understand that we may ask. |
| 17:17.21 | [TK]D-Fender | drfreeze, And when you do your echo test... ANSWER the call first. Never just call Echo() right away |
| 17:17.47 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:19.41 | p3nguin | echo test: http://pastebin.com/2fmtHqSQ |
| 17:20.37 | [TK]D-Fender | p3nguin, * would actually set up audio unless you answer the call first.... |
| 17:20.40 | [TK]D-Fender | won't* |
| 17:20.48 | p3nguin | Playback() does that. |
| 17:20.49 | [TK]D-Fender | s/actually/usually/ |
| 17:20.51 | [TK]D-Fender | gah... |
| 17:21.03 | [TK]D-Fender | Yes, Playback would do it |
| 17:22.50 | p3nguin | valken: As far as usage, applications can be run in extensions. Example: exten => 500,1,NoOp(); this is the NoOp() application. Functions are not executed like that. Functions are treated more like variables: Set(TIMEOUT(digit)=4); this uses the Set() application to write the value of '4' to the TIMEOUT function. |
| 17:24.18 | Valken | oh... |
| 17:24.32 | Valken | anyway, hope you didn't get offended |
| 17:24.43 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 17:24.57 | p3nguin | Functions can also be read like variables: Verbose(0,${TIMEOUT(digit)}); this uses the Verbose() application to print the value of function TIMEOUT(digit). |
| 17:25.09 | p3nguin | I'm not offended at all. |
| 17:25.40 | Valken | is the ael extension file popular? |
| 17:25.40 | p3nguin | I was just letting you know it was an application and not a function. |
| 17:25.51 | p3nguin | moderately |
| 17:25.59 | Valken | I did learned it more than a year ago, loved it, but for practical reason, I had to remain with .conf |
| 17:26.12 | p3nguin | There are some weird people who like to write extensions in AEL. |
| 17:26.23 | Valken | if the market do not use the ael, I wont either. |
| 17:26.30 | Valken | is weird |
| 17:26.41 | p3nguin | I think AEL has more limits than standard .conf does. |
| 17:26.59 | Valken | really? |
| 17:27.12 | leifmadsen | yes |
| 17:27.23 | leifmadsen | AEL is 'compiled' into dialplan |
| 17:27.25 | Valken | when I learned it, it looked so easy to do the imbricated if |
| 17:27.40 | leifmadsen | so AEL only supports what it can interpret |
| 17:28.08 | leifmadsen | AEL is also not readily maintained |
| 17:28.12 | Valken | ok |
| 17:28.15 | Valken | thank for the info. |
| 17:28.26 | leifmadsen | that's why we don't cover it at all in the Asterisk book |
| 17:28.42 | Valken | leifmadsen: which asterisk book (I'm looking for one) |
| 17:28.48 | leifmadsen | www.asteriskdocs.org |
| 17:28.58 | leifmadsen | pretty much the one you'll find if you google for "asterisk book" :) |
| 17:29.01 | Valken | ok |
| 17:29.08 | Valken | that's the one I'm in now |
| 17:29.17 | Valken | just not easy to take with me. |
| 17:30.29 | p3nguin | You can purchase a hard copy of it and get a PDF to boot. |
| 17:31.36 | Valken | no it's ok. The single html is good enough |
| 17:31.41 | Valken | I made a mistake with another project. |
| 17:34.58 | *** join/#asterisk carloimperia (~carloimpe@151.16.176.190) |
| 17:35.59 | leifmadsen | if it's not easy to take, get a hard copy and support the authors. If that isn't necessary, then don't complain it's non-portable |
| 17:39.29 | Gugge | i like AEL, but some things is just strange in AEL :) |
| 17:41.43 | drfreeze | When playing a sound file the sound is garbled |
| 17:41.55 | drfreeze | When it gets to the echo portion, the call hangs up |
| 17:42.56 | *** join/#asterisk hdiogenes (~humberto@189.124.200.92) |
| 17:42.58 | p3nguin | How about that SIP debug that was requested? |
| 18:02.48 | jaytee | I bought the book and got a PDF version but I don't remember getting a boot? |
| 18:04.35 | p3nguin | "to boot," not "and a boot." :) |
| 18:05.07 | p3nguin | Why would you want just one boot, anyway? |
| 18:05.11 | *** join/#asterisk ermali82 (~eri82@213.207.42.45) |
| 18:05.22 | p3nguin | You'd have to buy two copies of the book to get a pair for wearing. |
| 18:10.06 | ermali82 | does anyone know what variable is filled with the value of UUI in the dialplan? I'm trying to get the value of ${USERUSERINFO} but it show empty |
| 18:10.46 | leifmadsen | what is UUI? |
| 18:11.07 | ermali82 | user-to-user-info |
| 18:11.13 | leifmadsen | does asterisk actually support that? |
| 18:11.16 | ermali82 | from pri line |
| 18:11.19 | ermali82 | yes |
| 18:11.20 | leifmadsen | I've never heard of it |
| 18:11.30 | leifmadsen | use DumpChan() to get the variable available on the channel |
| 18:11.41 | leifmadsen | or look at options to CHANNEL() function |
| 18:11.42 | ermali82 | it looks like it is broken |
| 18:12.05 | ermali82 | tryed DumpChan but no sign |
| 18:13.20 | leifmadsen | I've never heard of that at all |
| 18:13.22 | leifmadsen | so no idea |
| 18:13.37 | ermali82 | leifmadsen thanks |
| 18:14.11 | leifmadsen | do you have libpri loaded? |
| 18:14.49 | ermali82 | yes it is loaded and i se the value of the User-User-info |
| 18:15.04 | ermali82 | but cant set it to a variable |
| 18:15.22 | ermali82 | it shows the value of dump_user_user() |
| 18:16.08 | *** join/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 18:16.09 | leifmadsen | ermali82: look at line 57 of channel/sig_pri.c |
| 18:16.10 | leifmadsen | #undef SUPPORT_USERUSER |
| 18:16.19 | leifmadsen | not enabled by default |
| 18:16.21 | ermali82 | it is defined :( |
| 18:16.26 | ermali82 | i know |
| 18:16.30 | leifmadsen | I don't know... probably incredibly old |
| 18:16.37 | leifmadsen | that code path perhaps isn't followed anymore |
| 18:17.00 | ermali82 | the function pbx_builtin_setvar_helper set the value of variables? |
| 18:17.58 | leifmadsen | yes |
| 18:18.03 | leifmadsen | assuming that code gets called |
| 18:18.21 | ermali82 | have to try |
| 18:18.53 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 18:26.54 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 18:34.24 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:48.23 | *** join/#asterisk blizzow (~jburns@67.50.165.60) |
| 18:48.59 | _Corey_ | I've got a question for you guys... |
| 18:49.01 | _Corey_ | I had an inquiry about the "tdd mode" AGI function and whether it actually works... Has anyone had any experience with this for the deaf? Any tips or suggestions on better approaches? |
| 19:01.25 | wonderworld | hi, i am trying to playback a soundfile to both call legs after Dial() has bridged the legs. someone yesterday pointed me to using a macro. i wrote a macro just, playing back the file and triggered it with the U option in Dial() |
| 19:01.36 | wonderworld | sadly, no sound is playing for either party |
| 19:01.45 | wonderworld | what would be the correct way to do this? |
| 19:02.21 | p3nguin | Macro isn't trigged with U, it is triggered with M. |
| 19:02.37 | p3nguin | Gosub is triggered by U. |
| 19:03.04 | p3nguin | Pastebin the dial plan you are having problems with. |
| 19:03.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:03.07 | wonderworld | yeah you are right sorry...GoSub was what i ment. i tried macro before and i couldn't make it work either |
| 19:04.35 | *** join/#asterisk radic (~radic@dslb-094-216-253-120.pools.arcor-ip.net) |
| 19:04.41 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:05.03 | wonderworld | p3nguin: http://pastebin.com/Cxrur023 |
| 19:05.23 | wonderworld | i just want caller and callee to hear the same soundfile after Dial connects them |
| 19:05.47 | p3nguin | Are "1" and "2" valid audio files? |
| 19:05.54 | wonderworld | yes, they work fine |
| 19:06.02 | wonderworld | i tested them before |
| 19:06.15 | p3nguin | r,U(playfile) <--- syntax error |
| 19:06.19 | leifmadsen | wonderworld: you probably want G() option |
| 19:06.24 | p3nguin | extraneous comma |
| 19:06.29 | leifmadsen | and that |
| 19:07.16 | wonderworld | doh. haha. |
| 19:07.21 | wonderworld | let me try it without that comma |
| 19:09.10 | p3nguin | Also, you can consolidate setting CALLERID(num) and CALLERID(name) with Set(CALLERID(all)=Name Here <number here>) |
| 19:09.59 | p3nguin | Set(CALLERID(all)=Jack Frost <3149691077>) |
| 19:15.58 | *** join/#asterisk gusto (~gusto@nrbg-d9323bee.pool.mediaWays.net) |
| 19:16.03 | gusto | hi |
| 19:16.31 | gusto | i have this linksys pap2t-eu and somehow my analogue telephone that i connected to it aquired the time |
| 19:16.49 | gusto | i have no idea where it did get it from ... asterisk, or linksys fw? |
| 19:22.27 | p3nguin | The ATA support time servers, so maybe it comes from the ATA. |
| 19:22.41 | *** join/#asterisk fullstop (~quassel@64-121-54-104.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 19:23.17 | fullstop | Hi all. I'm using asterisk 1.6.xx in a particular solution, looking into sip and rtp packet tagging. |
| 19:23.30 | p3nguin | sip.conf |
| 19:24.13 | fullstop | The ip-qos.txt is not present in docs for 1.6... |
| 19:24.22 | gusto | ok, i fixed it, it was an adjustment on pap2t, but my telephone did not take it over, but that is OK, i did adjust that manually as well |
| 19:24.44 | fullstop | so, I was wondering, is there a default setting or if packets would be untagged unless it is set. |
| 19:24.51 | gusto | p3nguin: yes, i already found it |
| 19:26.35 | *** join/#asterisk gonewage (~gonewage@75-145-167-234-Illinois.hfc.comcastbusiness.net) |
| 19:26.42 | gusto | this pap2t is really cool |
| 19:26.48 | gusto | and it is smaller than i thought |
| 19:30.25 | gusto | i did two things in the meantime - i did test calls from ata/analog phone to ekiga and that worked very well, that made me happy |
| 19:30.54 | gusto | and then i subscribed to some prepaid providers so that i get some more experiences with sip |
| 19:31.41 | gusto | all this prepaid providers were easy to configure and work as expected, but i was not yet able to make calls because i have not filled them up, but that will happen soon (tomorrow) |
| 19:31.54 | gusto | and i ordered a book about asterisk (oreilly) |
| 19:31.58 | gusto | so ... |
| 19:32.41 | gusto | i put out heavy ordnance |
| 19:37.33 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:42.44 | *** join/#asterisk zeshoem (~zee@108.162.156.19) |
| 20:20.42 | gusto | btw. this hw version 0.3.5 means what? 3.0 16 MB 8 MB 2 Power adapter is 12V DC 1.0A. Marketed as the PAP2T.? |
| 20:22.23 | gusto | ah ... vs3 |
| 20:28.42 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:40.16 | *** join/#asterisk din3sh (din3sh@41.136.242.91) |
| 20:41.58 | *** join/#asterisk WindBack (~quassel@190.220.135.165) |
| 20:47.55 | gusto | i get |
| 20:47.57 | gusto | SIP/2.0 400 Bad Request-URI |
| 20:48.00 | gusto | on invite |
| 20:48.19 | *** join/#asterisk Valken (~valken@gw.ptr-80-238-173-103.customer.ch.netstream.com) |
| 20:48.38 | WindBack | Is there any way to send NOTIFY with message "waiting: yes" or message "waiting: no" from de dialplan. I'm developing a extension mobility feature from de dialplan and I want to send these message to alert the phone that user logged in |
| 20:48.42 | WindBack | ? |
| 20:49.24 | *** join/#asterisk angryuser_laptop (~angryuser@2a01:e34:ee09:8850:d1ef:90d8:d82:d97f) |
| 20:49.30 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 20:49.47 | p3nguin | gusto: Pastebin your dial plan. |
| 20:50.27 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 20:54.29 | gusto | http://pastebin.com/CbhjyfyY p3nguin |
| 20:59.23 | bobb_WU | can someone help me figure out why "modprobe wctdm24xxp" is not working? it says the module can't be found |
| 20:59.32 | bobb_WU | i just recompiled it (using emerge- i'm on gentoo) |
| 21:00.05 | gusto | well, and are you using that kernel? |
| 21:00.49 | gusto | i once also compiled a new kernel and then wondered why some modules did not load into, but that was because i forgot to boot into :-D |
| 21:01.25 | bobb_WU | yeah i did just update the kernel |
| 21:01.27 | bobb_WU | hmm |
| 21:01.56 | bobb_WU | how'd you get dahdi to recompile on the newer kernel? i remember having to do this once before now that you mention it |
| 21:02.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:06.11 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:07.12 | bobb_WU | no ideas? |
| 21:08.37 | bobb_WU | alrighty system upgrades are rocking and its past quitting time. thanks for the help, cya everyone tomorrow |
| 21:10.31 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 21:10.37 | paolosupino | Hi |
| 21:11.15 | paolosupino | How do I reset a voicemail to it's initial state? |
| 21:15.17 | leifmadsen | initial state? |
| 21:28.13 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 21:28.58 | paolosupino | leifmadsen: initial state as just after installing asterisk. |
| 21:29.24 | [TK]D-Fender | There is no voicemail as installed |
| 21:30.19 | [TK]D-Fender | Want to wipe a box? Delete the folder. |
| 21:35.31 | *** join/#asterisk gonewage (~gonewage@75-145-167-233-Illinois.hfc.comcastbusiness.net) |
| 21:38.20 | paolosupino | [TK]D-Fender: will wiping /var/spool/asterisk/voicemail also reset the passwords I changed when playing with voicemail? |
| 21:39.35 | [TK]D-Fender | paolosupino: No, that is voicemail.conf |
| 21:40.08 | leifmadsen | paolosupino: no, that is stored in voicemail.conf |
| 21:41.22 | paolosupino | [TK]D-Fender/leifmadsen: strange because in voicemail.conf I see the initial password I set for the voicemail, but if I try to use that password to login into voicemail it rejects it… |
| 21:41.51 | leifmadsen | are you using realtime voicemail then? |
| 21:42.10 | leifmadsen | anyways, for just basic voicemail, passwords are stored in voicemail.conf |
| 21:42.11 | [TK]D-Fender | paolosupino: And we see neither your VM configs nor the failed attempt to pick them up, etc |
| 21:42.21 | leifmadsen | that's where asterisk gets the password unless you setup a script for exterpass |
| 21:42.24 | [TK]D-Fender | ~pb |
| 21:42.24 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 21:42.25 | [TK]D-Fender | ^^^ |
| 21:44.03 | paolosupino | [TK]D-Fender: obviously the next step is to paste bin the configs… :-) Let me collect them and I will post the URL shortly…. |
| 21:46.52 | Kobaz | or else |
| 21:54.48 | paolosupino | [TK]D-Fender/leifmadsen: sanitizing the voicemail.conf file I found the answer… I have the setting password location=spooldir which creates the file secret.conf in the user's voicemail dir… |
| 21:55.01 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:55.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:59.21 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 21:59.29 | gusto | no one an idea? |
| 22:00.16 | Valken | gusto, what for? |
| 22:02.22 | fprior | [TK]D-Fender, Hi, did you remember spa400 issue we have discuss last week ? If yes, I'm turn sip debug on and I'm ready to start any test you can suggest to me (if you have time...) |
| 22:03.51 | gusto | Valken: http://pastebin.com/2xUqS0c1 |
| 22:04.14 | Valken | And what is the problem? |
| 22:04.26 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 22:04.27 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 22:04.45 | asteriskmonkey | had anyone had issues compiling asterisk with dahdi under freebsd in a jail? |
| 22:07.09 | gusto | Valken: i do not know, when i call an outside number it gets off |
| 22:07.16 | gusto | Valken: does not ring |
| 22:09.02 | bn-7bc | I am struggling with a strange problem, when a call comes in from pstn (via my itsp) I have perfeect 2 way sound but when I originate a call from a sip client to pstn here is no sound in amy direction |
| 22:09.04 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 22:10.08 | bn-7bc | diagram itsp -> Asterisk server -> nat device - sip client |
| 22:10.31 | bn-7bc | any ideas |
| 22:10.50 | gusto | i am not even that far, but it seems to have something to do with the codecs |
| 22:11.01 | gusto | maybe you should restrict it to some codecs |
| 22:11.17 | gusto | i did never make a call outside |
| 22:11.33 | gusto | so no voip provider works to me yet |
| 22:11.35 | Valken | gusto: is your firewall rtp ports open? |
| 22:11.38 | Valken | are |
| 22:11.43 | gusto | rtp??? |
| 22:11.46 | gusto | nooo |
| 22:11.51 | gusto | what is rtp port? |
| 22:11.52 | bn-7bc | gusto: already done that the client is restricted tu ulaw |
| 22:12.04 | Valken | sip only handle the connection but the audio goes over rtp |
| 22:12.17 | gusto | Valken: do you mean 5060? |
| 22:12.38 | Valken | you can check on /etc/asterisk/rtp.conf |
| 22:12.41 | Valken | no, 5060 is sip default port |
| 22:12.56 | Valken | by default, asterisk need ftp 10000 to 20000 opened |
| 22:13.05 | Valken | at least in 1.6 |
| 22:13.16 | gusto | rtpstart=10000 |
| 22:13.16 | gusto | rtpend=20000 |
| 22:13.28 | Valken | but normally, ringing should work even without rtp. |
| 22:13.34 | gusto | ys |
| 22:13.40 | gusto | yes .. you see! |
| 22:14.13 | bn-7bc | well but this still foes not explain why I have sound ven i get calls from pstn put not when i make outbound calls or am I missing something |
| 22:15.41 | gusto | i have great sound everytime |
| 22:15.47 | gusto | but to me only local calls work |
| 22:15.58 | gusto | but the quality is impressive |
| 22:19.19 | bn-7bc | uk just so I make myself 100% clear calls from pstn works great (to way sound call from cline to asterisk (to voicemail) also have 2 way song (i her dyhe menu and can also record new greetings |
| 22:19.28 | Valken | I'm not good at telephony but, have you checked that you are or not behind a firewall? |
| 22:19.59 | bn-7bc | outbound calls from same client vioa asterisk no sound in any direction ( |
| 22:20.51 | gusto | i am ON a firewall |
| 22:21.08 | bn-7bc | i am behind i firewall but shold hat not kill sonde in all cases that is for both inn an outbound calls? |
| 22:21.12 | gusto | my asterisk runs on my wndr3800 router with openwrt |
| 22:21.22 | gusto | and it has iptables firewall for NAT and so on |
| 22:21.26 | Valken | It's time to go to sleep. |
| 22:21.28 | Valken | Bye |
| 22:21.43 | bn-7bc | mine runs on a vps hosted by ipeer |
| 22:22.14 | gusto | hmm |
| 22:22.38 | gusto | could do that as well, i too have a vserver, but that is not interesting, because i need it locally for my ata adapter now |
| 22:22.50 | gusto | so that i do not need to cope with NAT and so on |
| 22:24.31 | gusto | so i have 2 messages, one good and one bad ... the good one is that asterisk18 and the ata adapter with my analogue telephone, all great, ... the bad message is that i still can not call anyone outside, it works only locally (so nothing new) |
| 22:25.09 | bn-7bc | ok, I'm kind of in the middle of nowhere now i will try over 3g when i get back to a covered area to steer clear of nat altogether (yes we still get routable addresses on 3g here in sweeden |
| 22:26.46 | gusto | you see |
| 22:27.07 | bn-7bc | I'll research a bit more tomorrow but now it's time for bed (tip: never table stance voip problems last thing before bed) |
| 22:27.16 | gusto | here in germany all providers (tmobile, vodafone, and o2) do give you NATed addresses over 3g |
| 22:28.12 | bn-7bc | We all look forward to IPV& when this nat nonsense will be gone |
| 22:28.31 | bn-7bc | hates nat with a passion |
| 22:28.43 | gusto | its IPV^ |
| 22:28.47 | gusto | not IPV& |
| 22:28.49 | gusto | :-D |
| 22:28.58 | *** join/#asterisk jsjc (~Adium@138.Red-83-60-133.dynamicIP.rima-tde.net) |
| 22:29.17 | gusto | NAT is not bad, it just does not need to be around |
| 22:30.37 | bn-7bc | ipv6 |
| 22:32.13 | gusto | jj |
| 22:32.14 | bn-7bc | well nat sucs with protocols that us ip addresses in the payload (paging sip) |
| 22:32.19 | gusto | i am going to sleep as well |
| 22:32.41 | gusto | i have a book about IPv6, maybe that will help to get sleep |
| 22:37.27 | bn-7bc | well imho ipv& is mostly just IPV4 with larger address spac a few header tweaks and stander sybnet side of 64 bits |
| 22:38.05 | bn-7bc | + dhcp only needed (at least in home networks) for dns addresses |
| 22:39.54 | p3nguin | <Valken> by default, asterisk need ftp 10000 to 20000 opened What? FTP? |
| 22:41.06 | bn-7bc | oh yes and a few changes to firewalls (esp what ever you do don't ever block all icmp pacts from van as this will break connectivity to devices with lower path mtu the you) |
| 22:41.39 | bn-7bc | he meant ftp port 10000-20000 |
| 22:42.21 | p3nguin | What?! FTP? |
| 22:42.37 | p3nguin | You people are on crack today. |
| 22:42.39 | bn-7bc | well of to bed wil heck tomorrow |
| 22:42.52 | bn-7bc | no asterisk |
| 22:43.09 | bn-7bc | ftp was a typo for rtp |
| 23:01.26 | fprior | Hi, Is there any way to tell Asterisk not to generate additional headers X-Asterisk-HangupCause and X-Asterisk-HangupCauseCode ? |
| 23:14.08 | blizzow | Is there a way to tell what may have cause my asterisk daemon to either get reloaded or crash and reload on it's own? I'm looking in /var/log/asterisk/full and I see where it starts parsing /etc/asterisk/codecs.conf and begins parsing the rest of the config, but I don't see anything that gives me any diagnostic issue as to what started the entire reload process. |
| 23:14.48 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:15.58 | p3nguin | its own |
| 23:19.18 | *** join/#asterisk martinintn (~martindco@adsl-74-179-230-213.bna.bellsouth.net) |
| 23:20.26 | blizzow | sorry p3nguin, good catch. I apparently need more caffeine. |
| 23:34.25 | *** join/#asterisk newbchessplayer (~newbchess@unaffiliated/newbchessplayer) |
| 23:36.07 | paolosupino | please take a look at: http://pastebin.com/KZbCd0fZ |
| 23:37.28 | paolosupino | I have the following problem: it used to work (giving the right caller ID), but now (and voipcheap are saying they didn't change anything) it either gives me a private number or a number that is completely unrelated to the callerid(num) that I set in extensions.conf. |
| 23:37.41 | paolosupino | Can anyone spot where I'm doing something wrong? |
| 23:37.49 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 23:38.41 | p3nguin | Dial(SIP/voipcheap/${EXTEN}) |
| 23:38.53 | p3nguin | sendrpid=yes |
| 23:40.22 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 23:45.13 | paolosupino | p3nguin: same inconstant behavior… |
| 23:45.27 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 23:50.45 | drfreeze | Anyone know how to get a hangup event when checking voicemail? |
| 23:52.32 | *** join/#asterisk albertoandrade (~albertoan@186.206.5.67) |
| 23:59.08 | p3nguin | paolosupino: Need a sip debug of a call where the problem is happening. |