| 00:00.28 | [TK]D-Fender | you need to prove that softphone. |
| 00:00.39 | [TK]D-Fender | basically that's the wildacrd of the bunch. |
| 00:03.43 | ChannelZ-Wk | Pretty sure you just have audio device/mixer problems. |
| 00:12.25 | *** join/#asterisk areski (~areski@80.174.128.116.dyn.user.ono.com) |
| 00:26.57 | WanDang | any tricks you can suggest to prove the softphone? Given that the mike produces audio in the speakers whether or not I'm using the softphone (whether or not it is even running on the computer), so it's a bit tricky to tell if echo is working or if it's just the normal audio. Any suggestions? |
| 00:29.02 | *** join/#asterisk FedoraUser (~FedoraUse@fedora/FedoraUser) |
| 00:29.06 | FedoraUser | hi friends |
| 00:29.36 | FedoraUser | I am new to asterisk and I managed to create server for my family |
| 00:29.43 | FedoraUser | I am using it with linphone |
| 00:30.13 | FedoraUser | I cannot add users by user@domain, and when I call the extension, i see that user is actually user@serverip |
| 00:30.30 | FedoraUser | is it easy to fix, any pointers? |
| 00:30.49 | FedoraUser | I have trouble forming google query, that's why I came here |
| 00:34.34 | newtonr | First, try zoiper, as I remember linphone being terrible and flakey |
| 00:34.54 | newtonr | Second, where are you "adding users"? |
| 00:35.30 | FedoraUser | newtonr, thanks, extensions.conf |
| 00:36.00 | FedoraUser | newtonr, I used linphone because of huge multiplatform support, I will look into zoiper |
| 00:36.24 | newtonr | Usually in a sip phone, you'll add a user like 6001@somehostIPorDomain |
| 00:37.05 | newtonr | when the phone calls Asterisk it should look for 6001 in the context (extensions.conf) defined in the configuration for the SIP account in the channel driver config (sip.conf or pjsip.conf) |
| 00:37.45 | newtonr | FedoraUser, probably easiest to look at this simple tutorial that uses zoiper https://wiki.asterisk.org/wiki/display/AST/Hello+World |
| 00:38.06 | FedoraUser | newtonr, oh, my bad - I did define user in sip.conf and then added to extensions.conf |
| 00:38.23 | FedoraUser | newtonr, ooooh, that's what I looked for! thanks man - appreciate it |
| 00:38.39 | newtonr | extensions.conf is your Dialplan that is where you decide what to do with a call. In sip.conf you are configuring Asterisk's connectivity to a SIP device. |
| 00:39.11 | newtonr | the configuration in sip.conf will determine where Asterisk starts executing dialplan in regards to a new call/channel. |
| 00:39.58 | FedoraUser | newtonr, thanks a lot, I will read up on Hello World and look into sample sip.conf to figure out where to put my fqdn |
| 00:40.15 | newtonr | No problem |
| 00:40.30 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Architecture%2C+The+Big+Picture this page will help you get an idea about all the parts and pieces involved so you may learn more of what to search for on the wiki. |
| 00:40.40 | [TK]D-Fender | Normaly you enter the IP/host of your server into an account record and you never have to specify it again. |
| 00:40.53 | [TK]D-Fender | You'd just dial the # and the call would just use that account. |
| 00:42.33 | FedoraUser | [TK]D-Fender, I wanted to add my brother to a friend list, but couldn't find him using user@domain, that is what I am solving, otherwise calling works ;) |
| 00:42.44 | FedoraUser | anyway, thank you both for helping words |
| 00:42.49 | FedoraUser | off to research! :) |
| 00:44.10 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 00:48.53 | FedoraUser | zoiper is not FLOSS? |
| 00:50.36 | newtonr | I don't think so, but there is a free as in zero price version. |
| 00:50.43 | [TK]D-Fender | BEER |
| 00:51.22 | FedoraUser | ok, I really want to move away from proprietary solution, so I'll give linphone a shot, ty ;) |
| 00:51.25 | newtonr | [TK]D-Fender, I've had so many people confused by that analogy. |
| 00:52.09 | newtonr | FedoraUser, you might also try Jitsi. https://jitsi.org/Development/VersionControl |
| 00:52.54 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 00:53.08 | FedoraUser | newtonr, thanks! I tried jitsi, I have a lot of complaints with it, but imho it has bright future |
| 00:54.25 | newtonr | :) |
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| 01:04.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:39.11 | *** join/#asterisk Kattyroo (~Kattyroo@97-87-112-98.dhcp.stls.mo.charter.com) |
| 01:39.14 | Kattyroo | good evening, cupcakes! |
| 01:42.03 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:42.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:43.04 | WanDang | anyone tried sflphone? |
| 01:45.27 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
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| 02:17.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:31.03 | *** join/#asterisk cyford (~allen@76.122.73.37) |
| 02:32.02 | cyford | are there macro dumps for asterisk using pjsip like in extenstion custome sampe ? macro-dumpvars |
| 02:38.54 | Penguin | Did you try the regular DumpChan() application? |
| 02:41.41 | cyford | no, ill give it a shot |
| 02:41.47 | cyford | thanks |
| 02:43.26 | WanDang | for what it's worth, I just tried sflphone, and I have the same problem doing an echo test with it as I did ekiga phone. So it's not the softphone per se, probably, but something in how it's set up. I think ChannelZ-Wk may be right that there is an audio config prob |
| 02:45.41 | cyford | Penguin, it gave me info from the caller not the callie |
| 02:45.58 | Penguin | It gives channel info. |
| 02:46.46 | cyford | yes the callers channel |
| 02:47.29 | cyford | i dont even see the callies extenstion or codecs in it |
| 02:47.58 | cyford | opps nvm |
| 02:48.50 | ChannelZ | WanDang, Do you have a non-Linux machine to test with? Or even an Android phone you could install something like CSipSimple on? |
| 02:51.14 | WanDang | ChannelZ: no unfortunately. I do have an android phone, but there is no wifi on the lan where * and my client reside, so no way for android to connect to * |
| 02:52.14 | Penguin | I just installed csipsimple on my tablet, but haven't configured it yet. |
| 02:53.42 | cyford | i use xlite on tablet laptop, an zoper on phone and linux |
| 02:53.51 | ChannelZ | Well, I'm willing to bet your asterisk/DAHDI are fine and it's just the softphone you're grappling with.. which I don't know what to tell you there |
| 02:54.59 | ChannelZ | I could give you a test extension at my place you could call to do an echo test, via your analog->asterisk->sip to me.. with the added bonus of NAT happening |
| 03:00.42 | WanDang | ChannelZ: That would be cool! but there would be double nat happening. Right now, I'm inside the nat'ing firewall. I think I'd be better getting sip <-> * working by itself before adding nat'ing to the mix, don't you? |
| 03:01.10 | ChannelZ | Yes but it appears you don't have the means to do so :) |
| 03:01.26 | *** part/#asterisk FedoraUser (~FedoraUse@fedora/FedoraUser) |
| 03:01.47 | ChannelZ | Is your asterisk box behind NAT, or is its self the firewall? |
| 03:02.21 | WanDang | it's behind a natting firewall |
| 03:03.57 | ChannelZ | Well at some point you will have to setup a small range of ports to forward through it to your asterisk to get any external communication to work right |
| 03:04.09 | WanDang | the nat'ing firewall isn't directly on the Net either, tho. It gets a private IP (different subnet than the LAN) from the adsl device. It's the adsl device that has the public ip |
| 03:04.32 | WIMPy | Double NAT? |
| 03:05.26 | Penguin | If there's a DSL modem with a built-in router AND ALSO a router behind that doing NAT, that's what I like to call a configuration error. The modem should be in bridge mode if you have a router doing the NAT. |
| 03:05.29 | WanDang | WIMPy: each asterisk server is behind a NAT, so there is NAT going on at both ends of the connection |
| 03:06.05 | WIMPy | Sounds like three times NAT at the moment. |
| 03:06.44 | ChannelZ | Oh, there are 2 asterisks involved? |
| 03:07.11 | WanDang | channelZ: there would be two involved if I tried to echo test with your server |
| 03:07.18 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:07.36 | ChannelZ | Oh.. yeah.. ish - but my shit works :P |
| 03:07.58 | ChannelZ | My box isn't technically behind NAT |
| 03:08.03 | WIMPy | Famous last words? |
| 03:08.23 | Penguin | What does "technically" mean here? |
| 03:08.40 | WanDang | Penguin: I have the adsl device forwarding everything dmz style to my own nating router |
| 03:08.51 | Penguin | That's a configuration error. |
| 03:08.56 | WanDang | does that count as bridging |
| 03:09.01 | WanDang | oh |
| 03:09.03 | Penguin | Also known as "wrong." |
| 03:09.09 | Penguin | Put it in bridge mode. |
| 03:09.20 | Penguin | or stop using a second NAT router. |
| 03:09.28 | ChannelZ | It runs on the firewall. |
| 03:09.51 | Penguin | Is asterisk listening on the public address? |
| 03:09.52 | ChannelZ | So *I* am behind NAT but my asterisk is not. But I have another * box that is |
| 03:10.08 | ChannelZ | So I've got one of each |
| 03:10.39 | WanDang | well, I'm not currently trying to use SIP to use sip outside of the LAN, so I havn't really bothered with the the outside stuff yet. |
| 03:11.16 | Penguin | With or without SIP, you still wouldn't catch me setting up a network that way. |
| 03:11.59 | ChannelZ | Wear two condoms |
| 03:12.15 | Penguin | double-bagger |
| 03:14.19 | WanDang | ChannelZ: do you still think it's worth it to try and echo test to your asterisk server given that I appear to have configuration issues to the outside world? Afterall, like you said, i don't think there is a problemm with my * setup itself, it's probably something on the client |
| 03:14.52 | ChannelZ | Well |
| 03:14.56 | WIMPy | In your situation, I wouldn't even think of using sip. |
| 03:15.47 | Penguin | There's a slim chance that you have two NATs that play nicely with SIP/RTP. |
| 03:16.14 | ChannelZ | My system will pretty much ignore whatever IP your system *thinks* it is in favor of what it really is. So that part should work. It's just a matter of whether or not your inner-router will get the traffic back to your asterisk box correctly, of which I have doubts |
| 03:16.45 | [TK]D-Fender | Yup, you'll find those on the left-and side of the 2nd shelf to the back, right next to the unicorns :) |
| 03:18.39 | WIMPy | starts acampaign to free the unicors from [TK]D-Fenders shelve. |
| 03:18.56 | ChannelZ | I would say you would still probably want/have to port-forward a few ports from your inner router to your asterisk box, as set in rtp.conf (I think it defaults to a massive range but you can narrow it down to just a couple for the moment) |
| 03:24.40 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 03:27.38 | WanDang | I've already cut it down some. It defaults to 10000-20,000. I already limited that to 19,000-19,999. I guess I could limit it some more. I've been afraid to limit it too much because the o'reilly book says it takes "at least two" rtp port for each call, without saying howmany it is likely to take so I want to be sure to have enough available ports |
| 03:29.32 | ChannelZ | Well it takes 2, but only one incoming which is what rtp.conf is controlling. You need as many as you want simultaneous calls occuring on the box, plus a few pad. |
| 03:30.22 | ChannelZ | You've got plenty of ports to spare but I figure why have 1000 when I only need 10 |
| 03:30.27 | [TK]D-Fender | 2 per direction typically |
| 03:31.42 | WanDang | So, say I limit it to 10 ports in rtp.conf, and open those up on the inner natting firewall, should that be enough permit echo testing with your server? |
| 03:32.04 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 03:32.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:32.14 | WanDang | I guess I would have to port forward those ports from the nat box to the * box on my end? |
| 03:32.33 | ChannelZ | Yes |
| 03:33.22 | WanDang | sorry, I'm not sure which of my questions your answering yes to |
| 03:33.41 | ChannelZ | and ideally yes it would work, but hard to say. SIP can be a PITA even without NAT involved, much less multiple |
| 03:38.23 | ChannelZ | Yes to the port forwarding bit |
| 03:39.14 | WanDang | ChannelZ: it is getting late for me. Would you be williing to let me echo test against your server another day? because it is getting late here. I haven't done port forwarding with iptables for a couple of years, plus I want to followup on Penguin 's suggestion about putting the adsl device in bridge mode. Both those are going to require more documentation reading than I think I can keep myself up for |
| 03:39.51 | ChannelZ | Yeah no prob |
| 03:40.15 | [TK]D-Fender | I've seen many ADSL router/modem combo's where you can just PPPoE to it directly even if it is its own NAT and get another IP direct for your insider router |
| 03:41.37 | WanDang | ChannelZ: thanks. when are you likely to be back here next with a holiday coming up and all? |
| 03:42.30 | ChannelZ | I'll be here a little while tomorrow and then not back til later Thursday |
| 03:42.49 | WanDang | [TK]D-Fender: hmmm, guess I'll have a look at that too. |
| 03:43.00 | WanDang | ChannelZ: cool, thanks |
| 03:43.30 | WanDang | ChannelZ: you pacific time zone by any chance? |
| 03:43.31 | ChannelZ | Is Linux your inner NAT? |
| 03:43.51 | ChannelZ | or is it ADSL modemthingy->some routerthingy->asterisk |
| 03:44.03 | ChannelZ | I'm Mountain.. 8:43pm here currently |
| 03:44.16 | ChannelZ | whatever that is.. 1 or 2 hours ahead of pacific? I forget |
| 03:44.17 | WanDang | yes, my inner nat is a linux box |
| 03:44.33 | ChannelZ | ok that makes it easier |
| 03:44.54 | ChannelZ | (well, in that I know how to config it) |
| 03:45.16 | WIMPy | No need for port forwarding there. |
| 03:45.42 | ChannelZ | mmyeah |
| 03:46.30 | WanDang | ChannelZ: I will try to get my reading done for tomorrow before you sign off, if not, I will keep an eye out Thurdsay, thanks |
| 03:46.55 | ChannelZ | port forward is easy |
| 03:47.35 | ChannelZ | iptables -t nat -A PREROUTING --in-interface wan0 --protocol UDP --dport 10000:10100 --jump DNAT --to 192.168.2.2 # for example |
| 03:48.16 | ChannelZ | wan0 being whatever your external iface is, 10000:10100 being whatever ports you have in rtp.conf, and 192.168.2.2 being the IP of your * box |
| 03:49.07 | ChannelZ | you'd also need a normal rule on the FORWARD table if its default policy is set to drop/reject |
| 03:51.32 | WanDang | ya, that is ringing some bells. So, I'll try and find you here tomorrow to try this without putting my adsl device in bridge mode. I'm fading fast now and pretty sure I don't have enough steam for this tonight |
| 03:51.42 | Penguin | FORWARD is a chain in the filter table. |
| 03:51.55 | Penguin | Just sayin'. |
| 03:51.59 | Kattyroo | FORWARD is a button on the vcr |
| 03:52.05 | Kattyroo | Just sayin. |
| 03:52.14 | ChannelZ | on the whatnow? |
| 03:52.21 | ChannelZ | :) |
| 03:52.22 | WanDang | lol |
| 03:52.28 | WanDang | good night! |
| 03:52.32 | Kattyroo | glares at ChannelZ |
| 03:52.36 | Kattyroo | are you calling me old!? |
| 03:53.06 | ChannelZ | heh I'm old. And I work in TV |
| 03:53.12 | *** part/#asterisk WanDang (~Wang@24.140.224.212) |
| 03:53.46 | Kattyroo | pats ChannelZ on the back. |
| 03:53.51 | Kattyroo | guess we've both gotten old |
| 03:54.53 | ChannelZ | it happens |
| 03:57.59 | Kattyroo | nods grimmly |
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| 06:02.34 | cyford | should PJSIP_ENDPOINT(name,field) work in dialplan, i am getting No application 'PJSIP_ENDPOINT' for extension |
| 06:02.50 | cyford | asterisk 13 ofc |
| 06:07.25 | ChannelZ | Well, it's a function, not a dialplan app |
| 06:07.45 | cyford | thanks, just figuired it out lol |
| 06:30.41 | cyford | ok i almost have it working, only in pjsip it deletes some of the header and looks corrupt where in asterisk it displays the url fine ERROR: res_pjsip_registrar.c:504 rx_task: Unable to bind contact 'sip:3003@76.122.73.37:55451;transport=UDP;rinstance=f7934c5e8cf793ac;X-PUSH-URI=PJâM' to AOR '3003' |
| 06:33.04 | ChannelZ | what field are you querying? |
| 06:33.28 | cyford | exten => _300X,200,Set(location=${PJSIP_HEADER(read,X-PUSH-URI)}) |
| 06:34.03 | ChannelZ | oh different function |
| 06:34.47 | cyford | yeah i det error read_header: There was no header named X-PUSH-URI. |
| 06:34.57 | cyford | but i see it when registering |
| 06:35.04 | cyford | *Get |
| 06:35.31 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 06:35.57 | cyford | that first error unable to bind is when registering |
| 06:36.10 | cyford | but phone is registered |
| 06:36.46 | [TK]D-Fender | Taht only reads headers off an INVITE I'm betting, just like the chan_sip equivalent |
| 06:38.12 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:39.00 | [TK]D-Fender | Actaully... sorry, that is endpoint based |
| 06:39.15 | [TK]D-Fender | and I mentioned earlier that the contact is in the AOR, not the Endpoint |
| 06:39.38 | [TK]D-Fender | So it was probably a dead-end right off the start |
| 06:40.22 | [TK]D-Fender | And that's be in the "contact", not as a separate header I'm betting, just like the astdb chan_sip storage had it |
| 06:40.27 | [TK]D-Fender | it was not parsed out and separate |
| 06:43.00 | cyford | this looked simular to what i was trying to do , only just get info not update it exten => 1,1,Set(PJSIP_HEADER(add,X-MyHeader)=) |
| 06:43.10 | cyford | <PROTECTED> |
| 06:44.19 | [TK]D-Fender | cyford, You don't seem to be paying attention |
| 06:44.36 | [TK]D-Fender | cyford, that is not for inquiring about so OTHER communication that happened in the past |
| 06:44.46 | cyford | yes u say its in tha aor, |
| 06:44.52 | cyford | i read this too PJSIP_DIAL_CONTACTS(endpoint[,aor[,request_user]]) |
| 06:45.13 | cyford | i cant find a command to get it from there |
| 06:46.36 | cyford | when my phone registers aor is regecting the header, i think thts why my dailplan is failing |
| 06:47.27 | [TK]D-Fender | I'm not seeing any comparative attempts... |
| 06:47.49 | cyford | when i register i get this error Unable to bind contact 'sip:3003@76.122.73.37:55451;transport=UDP;rinstance=f7934c5e8cf793ac;M' to AOR '3003' There was no header named X-PUSH-URI.read_header: There was no header named X-PUSH-URI.X-PUSH-URI=PJ |
| 06:47.56 | [TK]D-Fender | You keep showing that... |
| 06:48.01 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:48.01 | [TK]D-Fender | and not a GOOD ONE |
| 06:48.08 | [TK]D-Fender | Or an attempt to pull for that AOR |
| 06:48.26 | cyford | if it cant bind it too aor how can i retrieve it |
| 06:48.36 | [TK]D-Fender | I'm not seeing yo trying. |
| 06:48.42 | [TK]D-Fender | Or confirming what a good one will even return |
| 06:49.59 | cyford | a good one will return something like this |
| 06:50.01 | cyford | 192.168.111.34:48574:3600:2002:sip:2002@192.168.111.34:48574;rinstance=1d02f625bec01f51;transport=UDP;X-PUSH-URI=http://s.notify.live.net/u/1/bn1/HmQAAACj_35LyORzuhjocOg5b2klKB4NHwhokx-yhaJUK25zTX8pjYsq-PmJV3DIVMq3PbyB5VlnZ-QBNzpacp9CnxO/d2luZG93c3Bob25lZGVmYXVsdA/EW_BnIu3fUOH7FePoWY"/var/lib/asterisk/agi-bin/zoiperpush.sh "192.168.111.34:48574:3600:2002:sip:2002@192.168.111.34:48574;rinsta |
| 06:50.01 | cyford | nce=1d02f625bec01f51;transport=UDP;X-PUSH-URI=http://s.notify.live.net/u/1/bn1/HmQAAACj_35LyORzuhjocOg5b2klKB4NHwhokx-yhaJUK25zTX8pjYsq-PmJV3DIVMq3PbyB5VlnZ-QBNzpacp9CnxO/d2luZG93c3Bob25lZGVmYXVsdA/EW_BnIu3fUOH7FePoWY |
| 06:50.09 | [TK]D-Fender | "will", or DOES? |
| 06:50.33 | [TK]D-Fender | Show us |
| 06:50.40 | cyford | it would, and when i use asterisk it does |
| 06:50.50 | [TK]D-Fender | Show us |
| 06:51.21 | [TK]D-Fender | CLI & dialplan to match |
| 06:51.22 | cyford | ok, its on my main box, 1 sec |
| 06:56.34 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 07:03.21 | cyford | ofc the cli an dialplan wont match |
| 07:03.38 | cyford | differnt complete versions and commands |
| 07:03.46 | [TK]D-Fender | ... I'm expecting a live sample right now.. so it should |
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| 07:10.03 | cyford | dam pc keep lockin up |
| 07:10.09 | cyford | brb |
| 07:15.44 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
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| 07:22.58 | *** join/#asterisk cyford (~allen@76.122.73.37) |
| 07:31.11 | cyford | ok whick verbose and debugging yu want see? |
| 07:31.38 | cyford | asterisk -vvvvvvv -g -dddddd -r ? |
| 07:33.56 | [TK]D-Fender | verbose 10 |
| 07:34.06 | [TK]D-Fender | no core required |
| 07:43.45 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 07:44.03 | cyford | http://pastebin.com/raw.php?i=mhUL5QTk |
| 07:44.20 | cyford | dialing is at the end.. |
| 07:44.48 | [TK]D-Fender | that is not at all what we were looking at |
| 07:45.08 | cyford | what u lokin for |
| 07:45.59 | cyford | it shows the push url included in the register, it shows the dial plan using it to pass it to the scrip and it shows the scrip success |
| 07:46.07 | [TK]D-Fender | We were talking about PJSIPAnd now 3/4 of an hour later you show chan_sip. |
| 07:46.08 | [TK]D-Fender | WTF |
| 07:46.18 | cyford | yes |
| 07:46.37 | cyford | i told you it wouldnt match up becuase 2 deffernt types |
| 07:46.55 | [TK]D-Fender | I said to show an ACTUAL TEST to prove what they even return |
| 07:47.13 | [TK]D-Fender | I could have built an entire * system in that time |
| 07:47.15 | cyford | ohh on the failling server? |
| 07:47.33 | cyford | ok 1 sec |
| 07:47.42 | cyford | <PROTECTED> |
| 07:47.47 | [TK]D-Fender | SHOW US WHAT THOSE FUNCTIONS EVEN RETURN TO PROVE THAT THIS ENTIRE LINE OF QUESTIONING HASN'T BEEN A COLLOSAL WASTE OF TIME. |
| 07:47.57 | [TK]D-Fender | (too late) |
| 07:48.36 | [TK]D-Fender | Looking at your chan_sip sample does absolutely nothing to validate if there is a workable PJSIP EQUIVALENT anywhere |
| 07:49.05 | cyford | thats becuase it ist using pjsip |
| 07:49.16 | cyford | if i had pjsip working i wouldnt be here |
| 07:49.46 | cyford | i had it working with asterisk sip, working on getting it working with pjsip |
| 07:49.58 | [TK]D-Fender | We're trying to prove if you're simply screwing up or not |
| 07:50.04 | [TK]D-Fender | And you're not SHOWING US |
| 07:50.14 | cyford | there is one error i think in my way |
| 07:50.18 | [TK]D-Fender | And pawning off OTHER code |
| 07:50.59 | cyford | i see the url entering pjsip , its just being truncated and rejected |
| 07:51.46 | cyford | i need to find out what is rejecting an why |
| 07:54.27 | *** join/#asterisk jagruti_ (~root@182.70.116.47) |
| 07:54.56 | jagruti_ | Hello |
| 07:55.41 | jagruti_ | I am getting connection failed error in asterisk. |
| 07:56.47 | [TK]D-Fender | jagruti_, Show us the exact message |
| 07:56.48 | jagruti_ | I am registering one a2billing user and dialing my mobile number. AGI is called but it is giving me error of Connection fail |
| 07:57.01 | [TK]D-Fender | ~pb |
| 07:57.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 07:57.03 | [TK]D-Fender | ^^^ |
| 07:57.27 | jagruti_ | http://forums.asterisk.org/viewtopic.php?f=1&t=92160 you can see here full details |
| 07:59.02 | [TK]D-Fender | We have no idea what that script is doing, and it is not something we support. You'll have to ask in an a2billing place |
| 08:00.32 | [TK]D-Fender | <PROTECTED> |
| 08:00.42 | jagruti_ | Is it any issue regarding asterisk? |
| 08:00.53 | [TK]D-Fender | no |
| 08:00.59 | [TK]D-Fender | This is a problem with that AGI |
| 08:01.12 | jagruti_ | okay Thanks |
| 08:01.53 | babak | Hi, is there reported problem about installing latest dahdi driver on ububtu 12.04 ? |
| 08:02.23 | [TK]D-Fender | babak, issues.asterisk.org <- go check the tracker |
| 08:03.21 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 08:07.20 | [TK]D-Fender | is done waiting for today |
| 08:13.32 | jagruti_ | Hey does anyone know what is the channel name for a2billing IRC? |
| 08:14.22 | cyford | i think that channel died a long timeago |
| 08:14.47 | jagruti_ | :( |
| 08:15.09 | cyford | i did like it though, an will be using it again after getting opensips |
| 08:15.18 | cyford | <PROTECTED> |
| 08:15.29 | jagruti_ | <PROTECTED> |
| 08:15.42 | jagruti_ | I am getting issue in AGI |
| 08:16.55 | cyford | did it say it support asterisk 8? |
| 08:17.19 | jagruti_ | cyford: sorry didn't get |
| 08:17.38 | cyford | i think i used asterisk 1.4 and 1.6 |
| 08:17.58 | jagruti_ | But in my other server this whole things working fine |
| 08:18.01 | cyford | if commands changed in asterisk that agi wouldnt work |
| 08:18.14 | cyford | same version asterisk? |
| 08:18.29 | jagruti_ | i have just make replica of that server to other server but getting this error |
| 08:19.18 | WIMPy | Well, obviousely not. |
| 08:19.34 | jagruti_ | ? |
| 08:20.02 | WIMPy | It it was a replica of a running server it would also run. |
| 08:20.09 | jagruti_ | WIMPy: Did you get why it is giving me error? |
| 08:20.19 | cyford | lol need sind cli agi debug on |
| 08:20.24 | cyford | *Send |
| 08:20.39 | WIMPy | I saw it's a2b and I can't comment on that. |
| 08:21.03 | jagruti_ | I have already show the CLI in this link http://forums.asterisk.org/viewtopic.php?f=1&t=92160 |
| 08:21.08 | cyford | i installed it ontop of freepbx in realtime modde |
| 08:22.27 | cyford | then i clustered an loadballanced it with central db , opensips and rsync |
| 08:22.27 | jagruti_ | I am stuck and can't figure out the issue |
| 08:26.02 | cyford | i need see more, debug everything, agi , sip and dial plan with verbose 10 |
| 08:26.32 | cyford | cant see the command that is failing |
| 08:26.42 | cyford | i also sent u a pm |
| 08:27.04 | WIMPy | Or just start over again getting a correct copy this time. |
| 08:28.45 | cyford | yeah mine was xenserver, snapshot then new vm 3 seconds |
| 08:31.29 | cyford | im also getting this error DEBUG[10811]: db.c:376 db_get_common: Unable to find key '3003;@sip:3003@76.122.73.37:58316;transport=UDP;rinstance=45d340b255edead1;X-PUSH-URI=h8' in family 'registrar/contact' |
| 08:35.08 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:43.41 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:50.15 | cyford | this seems to be my issue https://gitorious.org/asterisk-tools/asterisk/commit/f3d9f46ddce95aa2cd04de9f04dc2daabd088f8c |
| 09:02.59 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 09:05.21 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:20.41 | cyford | i think this may be my fix https://reviewboard.asterisk.org/r/3462/diff/ |
| 09:21.41 | *** join/#asterisk areski (~areski@80.174.128.116.dyn.user.ono.com) |
| 18:00.59 | *** join/#asterisk infobot (ibot@rikers.org) |
| 18:00.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:05.32 | ChannelZ-Wk | Johnny 5 is alive! |
| 18:06.57 | ChannelZ-Wk | No disassemble |
| 18:12.01 | *** join/#asterisk linetrace (~linetrace@65.19.81.126) |
| 18:31.34 | *** join/#asterisk infobot (ibot@rikers.org) |
| 18:31.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:31.35 | [TK]D-Fender | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 18:32.04 | pa | so there's no standard way to make a client re-register |
| 18:34.00 | rrittgarn | generally it is up to the phone to re-register, which generally happens at the half-life of the sip registration timeout |
| 18:34.29 | [TK]D-Fender | You don't make a phone register. Every |
| 18:34.33 | [TK]D-Fender | It DECIDES to. |
| 18:36.56 | pa | but i can lower the registration timeout right? |
| 18:46.36 | rrittgarn | you can configure the phone to want to register more often yes |
| 18:46.51 | rrittgarn | you can also use things like qualify to send sip options messages if you're having issues that are related to the far end firewall not keeping the port open |
| 18:46.54 | rrittgarn | (sometimes anyway) |
| 18:48.30 | Katty | UNINTENDED NACHOS |
| 18:53.30 | pa | rrittgarn, thanks, yes, actually that would do (setting the timeout on the client, that is luckily possible) |
| 18:54.03 | *** join/#asterisk jgowdy (data@irev.net) |
| 18:54.39 | jgowdy | In Asterisk 13, can you call ARI POST /channels/{channelId}/dtmf on a channel that is in a bridge? |
| 19:02.28 | ChannelZ-Wk | Aren't all channels bridges now? |
| 19:04.40 | mjordan | ChannelZ-Wk: nope |
| 19:05.25 | bsdice | <PROTECTED> |
| 19:07.40 | mjordan | jgowdy: yes, but the direction of DTMF is to the device. That being said, you may got some funkiness if the channel is in a bridge, as the audio coming from the device (IIRC) is dropped while we send DTMF to it |
| 19:08.01 | mjordan | bsdice: thanks, and np :-) |
| 19:09.29 | mjordan | ChannelZ-Wk: channels and bridges are still two separate and distinct objects/concepts/resources. However, many operations that previously could only be done to channels are now available on bridges - (a) bridges are accessible as a resource through ARI (and are generally not in dialplan/AGI/AMI) and (b) we made ways that those operations would work on a bridge in ARI |
| 19:11.22 | ChannelZ-Wk | Well yeah, bad phrasing on my part |
| 19:12.25 | jgowdy | mjordan: it's towards the external device you mean right? |
| 19:12.54 | ChannelZ-Wk | I guess I meant all channels _in_ bridges (for the purposes of sending dtmf, of the question) |
| 19:13.04 | jgowdy | so if I have a bridge, person with a phone on one channel, call out to some other IVR on the other channel, and I target the IVR channel and DTMF in theory |
| 19:13.20 | jgowdy | in theory, the third party IVR would hear that DTMF if the DTMF stars align? |
| 19:14.35 | jgowdy | I'm not trying to detect my own DTMFs, to drive my own IVR, I'm trying to push DTMFs for someone who is on the phone with say the American Express IVR |
| 19:15.16 | ChannelZ-Wk | beep beep boop |
| 19:16.56 | jgowdy | We're in Asterisk 11 land currently, and it's like, we can't ChannelRedirect away from Bridge, SendDTMF, and then re-bridge fast enough to even poorly have this functionality |
| 19:17.17 | jgowdy | And obviously calling SendDTMF on a channel locked in bridge deadlocks because threading model |
| 19:18.09 | jgowdy | ;win 2 |
| 19:18.47 | WIMPy | Why don't you chanspy()? |
| 19:20.22 | jgowdy | You can use chanspy to play DTMF onto a channel? |
| 19:21.20 | WIMPy | Why not? |
| 19:24.18 | ChannelZ-Wk | Well if you create your own channel and add yourself to the bridge couldn't you do anything you wanted? |
| 19:24.54 | WIMPy | That's the idea. |
| 19:37.17 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 19:39.22 | jgowdy | It depends on whether a 3 party bridge supports bridging DTMF |
| 19:39.33 | jgowdy | Because in the out of band case, it isn't just audio |
| 19:42.02 | jgowdy | And wouldn't the third channel that you add to the bridge, also be essentially "running bridge" |
| 19:42.05 | jgowdy | and thus be locked? |
| 19:43.20 | jgowdy | I don't see how you're going to be able to run SendDtmf on a channel that's participating in a bridge. That's the whole point of Stasis |
| 19:51.56 | goseeped | HI guys has anybody get video working on pjsip stack on webrtc support ? asterisk 13 seems to be rejecting vp8 sdp offer ! audio is fine ! |
| 19:53.41 | goseeped | any idea what could be happen ? |
| 19:58.28 | WIMPy | jgowdy: The same way you do with ChanSpy. You add a 3rd channel that connects to an application, or even a local channel. |
| 20:05.13 | goseeped | why astricon 2014 videos are private ? i couldn't go this year :S but i want to see what happen ! |
| 20:29.14 | mjordan | I don't see any AstriCon 2014 videos on the youtube channel |
| 20:29.59 | mjordan | jgowdy: are you implementing the IVR in Stasis/ARI? |
| 20:30.04 | *** join/#asterisk WanDang (~Wang@24.140.224.212) |
| 20:30.50 | WanDang | Hi, is ChannelZ still here today? |
| 20:31.21 | mjordan | see, here's the thing. If you're building an IVR in dialplan and you want to 'skip' to the next state of the IVR, you have to do something like that - emulate the DTMF for the user, etc. |
| 20:32.32 | mjordan | but if you're building an IVR in an external app - using Stasis/ARI - you don't need to do that |
| 20:32.50 | mjordan | if I've written an "IVR" in node.js, for example, and I know a user/phone should go to the next state, I can just change the state of the application |
| 20:34.05 | mjordan | I don't need to provide DTMF |
| 20:34.11 | mjordan | as my application is the full IVR. |
| 20:36.06 | *** join/#asterisk dundel (~dundel@190.98.82.5) |
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| 20:58.51 | ChannelZ-Wk | WanDang, yes |
| 21:00.33 | ChannelZ-Wk | though I'm about to run off to lunch and then will be home for a little bit before leaving again |
| 21:01.02 | WanDang | Hi, so it's maybe too late today to try some testing? |
| 21:01.41 | ChannelZ-Wk | No I can for a little bit in about an hour |
| 21:02.53 | WanDang | cool, thanks. ping me when you're ready. |
| 21:07.14 | rhineheart_m | hello |
| 21:08.43 | rhineheart_m | I have a problem with my sip server.. before I did not have issue with it.. it was running so well for more than 2 years. Apparently when I updated it, it not says Not detected by asterisk. Any suggestion? |
| 21:09.34 | *** join/#asterisk cyford (~allen@76.122.73.37) |
| 21:13.17 | [TK]D-Fender | What is "is" |
| 21:13.22 | [TK]D-Fender | "it"* |
| 21:15.11 | rhineheart_m | it is running elastix. I asked in their channel.. but I think no one is awake...seems every1 is asleep :) |
| 21:20.19 | [TK]D-Fender | You've given no detail whatsoever for your problem. |
| 21:20.27 | [TK]D-Fender | "it doesn't work" offers us nothing to go on. |
| 21:20.33 | [TK]D-Fender | "My car doesn't work, why?" |
| 21:21.02 | [TK]D-Fender | "it not says Not detected by asterisk" <- what is the "it" at the front of that question? |
| 21:21.25 | malcolmd | goseeped: the astricon 2014 videos have just been uploaded and tagged this week. the private tag, which has been in place while that's been happening, should come off next week |
| 21:21.57 | [TK]D-Fender | How can something else say that "it" is not detected by Asterisk? That would be 3rd parrty phrasing. |
| 21:27.25 | rhineheart_m | sorry for that. I am using atcom card http://www.atcom.cn/products_banka_mn94.html and I did update lately with one of the modules causing the issue. I forgot the exact name of the module. |
| 21:28.20 | [TK]D-Fender | You haven't properly described the issue at all yet. |
| 21:28.28 | [TK]D-Fender | what EXACTLY is not working? |
| 21:29.09 | rhineheart_m | when I do outgoing call it says all circuits are busy..try to call again later. |
| 21:29.19 | [TK]D-Fender | Show uas the failed call in a pastebin |
| 21:29.23 | [TK]D-Fender | ~pb |
| 21:29.23 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:29.24 | [TK]D-Fender | ^^^ |
| 21:30.51 | rhineheart_m | what logs do you ask? :) |
| 21:31.46 | [TK]D-Fender | Asterisk CLI |
| 21:31.50 | [TK]D-Fender | Logs = worthless |
| 21:31.59 | [TK]D-Fender | place a call. Pastebin the whole thing |
| 21:32.37 | WIMPy | So tonight is the time to say good bye to another great technological breakthough of it's time. This is the last day of Inmarsat-M. |
| 21:35.02 | *** join/#asterisk jvwjgames (~jvwjgames@c-174-52-179-61.hsd1.ut.comcast.net) |
| 21:35.09 | jvwjgames | HI |
| 21:41.41 | rhineheart_m | http://pastebin.com/shkXBf87 |
| 21:43.02 | [TK]D-Fender | [2015-01-01 05:37:10] WARNING[5058][C-00000009]: channel.c:6000 ast_request: No channel type registered for 'DAHDI' |
| 21:43.08 | [TK]D-Fender | chan_dahdi is not even laoded |
| 21:43.11 | [TK]D-Fender | loaded* |
| 21:43.25 | [TK]D-Fender | "module load chan_dahdi.so" <- do ti and PB the result |
| 21:43.38 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 21:43.48 | [TK]D-Fender | along with "dahdi show status", "dahdi show channels" following if it doesn't error out |
| 21:45.16 | rhineheart_m | http://pastebin.com/eSUgfrpC |
| 21:46.04 | [TK]D-Fender | Looks like you have a DAHDI & Asterisk version mismatch |
| 21:47.06 | rhineheart_m | http://pastebin.com/iGWkSQm6 |
| 21:48.09 | [TK]D-Fender | I said if it DIDN'T error out. |
| 21:48.11 | [TK]D-Fender | Which it did |
| 21:48.16 | [TK]D-Fender | So there's no point for those 2 |
| 21:48.26 | [TK]D-Fender | fix your package versions so they match. |
| 21:48.35 | [TK]D-Fender | "yum update" might do it if you're lucky. |
| 21:48.49 | rhineheart_m | :) |
| 21:48.50 | [TK]D-Fender | This is an elastix packaging issue |
| 21:48.55 | rhineheart_m | http://pastebin.com/xz42FhFe |
| 21:50.51 | [TK]D-Fender | I see a new set of Asterisk & DAHDI packages there. |
| 21:50.56 | [TK]D-Fender | take your updates and see |
| 21:51.39 | rhineheart_m | http://pastebin.com/AnRGt9JM |
| 21:52.36 | [TK]D-Fender | That doesn't appear to be the end of the update attempt |
| 21:54.34 | rhineheart_m | it tries to fetch glibc-devel-2.5-1 but returns an error I think |
| 21:55.05 | rhineheart_m | Error Downloading Packages: |
| 21:55.05 | rhineheart_m | <PROTECTED> |
| 21:55.57 | [TK]D-Fender | I'm still not seeing an end... |
| 21:59.27 | rhineheart_m | any more suggestion? |
| 21:59.37 | cyford | anyone know the location of <pjsip/sip_config.h> |
| 22:01.19 | ChannelZ | GRRR. The joy of leaving work early is immediately offset by being stuck on a freeway surrounded by a bunch of twats on their cell phones who can barely drive as it is, going half the speed limit |
| 22:01.39 | [TK]D-Fender | rhineheart_m, I don't see the END of the entire process. You pasting 1 tiny little line at at time and I don't see how it all ENDS. |
| 22:04.18 | ChannelZ | HOW WILL IT END!? |
| 22:05.38 | rhineheart_m | http://pastebin.com/LxBmA5sP |
| 22:06.14 | [TK]D-Fender | Why don't I see you reaching a prompt at the end of this? |
| 22:19.18 | rhineheart_m | http://pastebin.com/sV7emnAa |
| 22:22.07 | ChannelZ | network problems |
| 22:30.59 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 22:31.22 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:32.40 | rhineheart_m | I can wget the file @ http://mirror.pregi.net/pub/Linux/CentOS/5.11/os/x86_64/CentOS/glibc-devel-2.5-123.x86_64.rpm |
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