| 00:10.53 | [TK]D-Fender | because it is looking at the From for who is being contacted |
| 00:10.58 | [TK]D-Fender | and that doesn't auth inbound |
| 00:11.03 | [TK]D-Fender | that setts the OUTBOUND From |
| 00:13.12 | RadicalDev | Not sure I follow. What could I do so that I don't have to rewrite the fromuser on kamailio's side? |
| 00:14.31 | RadicalDev | Basically, the fromuser could be anybody, but I want to say that any INVITE coming from one of these kamailio servers that properly authenticate should be allowed to continue into the dialplan |
| 00:17.59 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:21.01 | [TK]D-Fender | then make a peer that actually matches the kamailio server |
| 00:21.13 | [TK]D-Fender | Which since the NAME isn't good.. it should match by IP... |
| 00:21.23 | [TK]D-Fender | when "dynamic" does NOT do for you |
| 00:21.27 | [TK]D-Fender | which* |
| 00:21.42 | [TK]D-Fender | it should be type= PEER and specify a host and not to challenge the call |
| 00:22.39 | RadicalDev | That's the thing, though. I don't want to have to care about the IP. There's no way I can say, "Look, challenge all invites, and if they provide a digest authorization username and response that's expected, let them through"? |
| 00:25.14 | RadicalDev | what else can I match on besides the source IP and caller ID? |
| 00:25.56 | [TK]D-Fender | it'll match on the from from first |
| 00:26.03 | [TK]D-Fender | you want something more advanced look at pjsip |
| 00:26.12 | [TK]D-Fender | chan_sip only goes so far |
| 00:26.23 | RadicalDev | I guess it's about that time, then =) |
| 00:26.27 | RadicalDev | thanks Fender |
| 00:26.49 | [TK]D-Fender | but sine you're allowing it through kamailio you seemed to think it should be trustworthy by default... |
| 00:27.02 | [TK]D-Fender | so then you should just allow that to be a proper peer and to match them |
| 00:27.54 | RadicalDev | Yeah, I'm redoing my dynamic scaling. Prior versions would do some really hacky things like use sip.conf templates and reload them on the fly. |
| 00:28.19 | RadicalDev | I'm going for stupid simple but also pretty secure |
| 00:29.02 | [TK]D-Fender | You could have that security moved the the kamailio layer and leave * dumb only to that level |
| 00:29.32 | RadicalDev | I've thought about that, but I like the idea of having to authenticate with asterisk as well as get past the firewall |
| 00:32.14 | *** join/#asterisk FuriousGeorge (~chatzilla@pool-71-127-243-42.nwrknj.fios.verizon.net) |
| 00:33.37 | [TK]D-Fender | Your call... |
| 00:49.10 | FuriousGeorge | http://pastebin.ca/3486386 |
| 00:49.52 | FuriousGeorge | maybe i am misreading that, but if /var/run/asterisk is owned by asterisk and in group asterisk, why would asterisk fail to create a pid? |
| 00:50.01 | diegoviola | any phone numbers I can call for testing? |
| 00:50.02 | FuriousGeorge | if i run it from the console it creates one |
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| 01:00.11 | lvlinux | Did Dial() used to use the pipe symbol | instead of comma to separate arguments? |
| 01:01.30 | rmudgett | lvlinux: Yes. The pipe symbol used to be used to separate arguments. That was changed after v1.4. |
| 01:02.20 | lvlinux | ah. K that's why I see it used strangely in varios dialplan code bits on the net. :-D Thanks |
| 01:02.21 | lvlinux | s/varios/various/g |
| 01:08.40 | FuriousGeorge | if i google search " Job for asterisk.service failed because a configured resource limit was exceeded" |
| 01:08.58 | FuriousGeorge | i get my own pastebin made 22 minutes ago :/ |
| 01:18.59 | diegoviola | http://thetestcall.blogspot.com.br/ <-- if I call this number internationally will I get billed for it? |
| 01:26.40 | *** join/#asterisk rekrul (ce1db035@gateway/web/freenode/ip.206.29.176.53) |
| 01:27.47 | rekrul | So, hoping someone can help me. I have asterisk install (latest LTS version). I seem to be able to register with my soft client fine but when I try to talk to my provider, I keep seing a ICMP error message of port unreachable on 5060. |
| 01:28.03 | rekrul | But, my soft client was able to connect just fine. |
| 01:29.07 | rekrul | I have verified I have no firewall rules blocking me, I am getting the messages all the way to the server, I'm just dropping them on teh floor and I don;t know why |
| 01:29.37 | rekrul | can anyone give me a hint as to what the issue might be? My google foo is failing me |
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| 01:36.59 | rekrul | Anyone????? |
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| 02:04.57 | ChannelZ | Was the softphone that worked running on the same machine as asterisk? |
| 02:05.09 | rekrul | no, different machine |
| 02:05.18 | ChannelZ | If no, is the asterisk machine its self acting as a firewall doing NAT for that computer it was on? |
| 02:05.33 | ChannelZ | If yes, can the firewall its self talk/receive on 5060 UDP? |
| 02:05.47 | rekrul | no, I have ensure that iptables and selinux is disabled |
| 02:05.52 | rekrul | I have verified this several times too |
| 02:06.12 | ChannelZ | Then are you sure asterisk is trying to contact the correct IP? |
| 02:06.22 | rekrul | it's always when I am getting SIP invites from my trunk too |
| 02:06.26 | ChannelZ | (it's not resolving DNS differently for instance) |
| 02:06.47 | ChannelZ | So you get the INVITE but then nothing happens? |
| 02:07.07 | rekrul | no, certain of that, verified that the DNS name resolves to the expected IP and see it trying to talk to the asterisk server, asterisk is responding with port unavaliable |
| 02:07.54 | rekrul | SIP invite comes in on port 5060, asterisk server replies destination port unreachable (centOS 6.7 server) |
| 02:08.18 | rekrul | I find this to be very very strange |
| 02:08.48 | rekrul | sorry, ICMP type 3 code 3 (destination port unreachable) |
| 02:08.57 | ChannelZ | Indeed. Sounds like a routing problem or otherwise being blocked |
| 02:09.47 | rekrul | but my SIP client is connecting to that very same server using the same IP address as the provider is talking to me on, netstat is 0.0.0.0:5060 for UDP |
| 02:10.07 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 02:10.28 | rekrul | if that were the case, why would the error message be comming from my server IP address |
| 02:11.37 | [TK]D-Fender | Show us the actual debug |
| 02:12.04 | ChannelZ | Because it can't route the packet anywhere? |
| 02:12.31 | rekrul | I need to sanetize the log file, will take a bit (real IPs) |
| 02:15.07 | [TK]D-Fender | Rule #1: do not screw with the evidence |
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| 02:17.11 | rekrul | fair enough, but I'm not comfortable shipping out real IPs to internet strangers...taht's not safe |
| 02:17.21 | rekrul | general rule I follw |
| 02:18.07 | rekrul | sorry, weeee bit paranoid |
| 02:21.18 | rekrul | well, one thing I do notice in the logs is this "Retransmitting #6 (NAT) to <provider IP>:5060", when I get a SIP response back (on port 5060), I am sending back to the provider port unreachable error message |
| 02:22.04 | rekrul | so, this is occuring when I initiate a call, I send the invite TO the provider, they try to respond with a TRYING message which is getting rejected |
| 02:23.19 | rekrul | Weird thing is that if I explicitly set nat=no, nothing comes back from the provider network at all |
| 02:24.07 | rekrul | did something change with nat default setting in asterisk? I thought it was defaulted to no unless explicitly set |
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| 02:27.13 | rekrul | I am also able to ping the IP address of the provider server just fine so it's not a routing issue |
| 02:27.48 | rekrul | they are also able to get to me just fine too. |
| 02:28.47 | rekrul | (I can see them trying to send me 100 TRYING messages (which I proceed to drop on the floor) |
| 02:28.49 | [TK]D-Fender | ping proves nothing |
| 02:28.55 | [TK]D-Fender | and we're not looking at the whole packet |
| 02:29.03 | [TK]D-Fender | that doesn't prove return addressing or anything else |
| 02:29.06 | rekrul | proves there isn;t a routing issue |
| 02:29.13 | [TK]D-Fender | and there are several other things possible we aren't seeing |
| 02:29.24 | rekrul | ok, such as? |
| 02:29.32 | [TK]D-Fender | there are MULTIPLE addresses in SIP |
| 02:29.42 | [TK]D-Fender | But I'm not going to play blind games with this |
| 02:29.52 | rekrul | ok |
| 02:30.02 | [TK]D-Fender | if I get real debug then I'll look at it |
| 02:30.54 | rekrul | well, given that I am seeing the provider return a 100 TRYING for the IP i expect it to come in on and I respond with an ICMP port unreachable error, I would think that is a pretty clear issue. No? |
| 02:32.00 | [TK]D-Fender | Nothing is clear when I'm not looking at complete packets myself |
| 02:32.20 | [TK]D-Fender | If you want to PW & PM it fine, but I'm not doing this blind |
| 02:33.58 | rekrul | 4816.066449000<providerIP><myIP>SIP370Status: 100 Trying | followed by 4916.066499000<myIP><provider>ICMP398Destination unreachable (Port unreachable) |
| 02:34.46 | rekrul | Sorry but I can't jeopardize my job like that. |
| 02:35.31 | [TK]D-Fender | You're looking at PACKET addresses and not PAYLOAD ADDRESSES |
| 02:36.00 | [TK]D-Fender | And like any mechanic we're not going to know what's wrong when you won't open the hood. |
| 02:37.28 | rekrul | hrmm, I don't know how best to handle this then |
| 02:37.36 | lorsungcu | pastebin the capture |
| 02:38.32 | rekrul | that's a good way to lose my job though |
| 02:38.43 | lorsungcu | if publicly routed addresses get leaked...publicly? |
| 02:38.54 | lorsungcu | on a psatebin that expires in 10 minutes? |
| 02:39.10 | rekrul | hold on a second...I just realzed, it's destined for port 24429 |
| 02:39.19 | lorsungcu | sounds like nat. |
| 02:39.26 | lorsungcu | sounds like this maybe isn't your day job |
| 02:39.48 | rekrul | It's other fields that I'm concerned about too |
| 02:40.20 | lorsungcu | well, good luck |
| 02:40.24 | rekrul | It's not my day job, learning a new skill on the job at the moment. I have messed with Asterisk in the past, new job |
| 02:40.37 | rekrul | role |
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| 06:30.04 | Lope | What's it called when you have a VoIP test extension where you can say a phrase, and the system will repeat it back to you? Is there a way to implement it easily in Asterisk? |
| 06:30.44 | Lope | And secondly, where I can press DTMF and it will say the tones back to me? |
| 06:31.57 | TandyUK | Lope: an echo service? |
| 06:32.22 | TandyUK | not sure about the second one |
| 06:32.22 | Lope | echo service yes |
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| 06:38.40 | ruben23 | hi there guys |
| 06:40.24 | ruben23 | anyone can help implement odbc to run a mysql cdr functions for asterisk calls somehow..we installed odbc already and setup mysql database and populate credentials on every config but i still getting this error -----> http://pastebin.com/QJZY5juc |
| 06:42.13 | ChannelZ | Lope, you'd have to do something like Read() and then SayDigits() |
| 06:43.06 | Lope | ChannelZ Okay, I've worked with those things before, I'll give it a go :) |
| 06:43.10 | Lope | ChannelZ: thanks |
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| 07:11.54 | ruben23 | any idea guys.. |
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| 08:17.06 | phpboy | /connect/quit |
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| 09:11.02 | Lope | anyone familiar with siproxy? |
| 09:11.57 | Lope | I tested it previously on a VPS with a dedicated IP. Now I'm trying to use it on my NAT router on my LAN where my softphone is. Having trouble connecting. |
| 09:12.19 | Lope | Having trouble with registration |
| 09:14.10 | Lope | Looks like I had not activated plugin_fix_bogus_via.so before. |
| 09:20.28 | Lope | I guess I'm gonna have to tcpdump it, because I've got no clue why it's not working. |
| 09:25.55 | Lope | Ah, this is a huge bummer: https://sourceforge.net/projects/siproxd/files/siproxd/0.8.1/ interoperability with callcentric issues. Fails with 500 Internal server Error. I got that issue :/ But then I had another issue with my other provider. |
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| 10:37.59 | gusto | hey |
| 10:38.10 | gusto | I have narrowed the problem down to chan_sip.so |
| 10:38.27 | gusto | I see that there are problems with the memory menagement on the udp receive and send buffer |
| 10:39.02 | gusto | when I make it too small, it drops calls, of course, bigger one does work, but too big one says that he can not create socket because there is NO ?buffer? |
| 10:39.24 | gusto | [Apr 13 12:34:08] WARNING[-1]: chan_sip.c:32200 reload_config: Unable to create SIP socket: No buffer space available |
| 10:39.42 | gusto | and that is bullshit, because there it was set to 1 MB or so I think |
| 10:40.32 | gusto | or maybe even 64 MB of udp buffer, yes, too much, I know, but the message is at least showing that he has problems in recognizing it |
| 10:43.14 | gusto | also interesting is that on openbsd I can disable udp checksums over sysctl but asterisk tells me that it is not supported by operating system |
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| 10:48.57 | gusto | ok, unbound filed as well, so maybe it is a problem at the openbsd kernel side |
| 10:49.01 | gusto | lol |
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| 10:51.57 | wyoung | hi gang |
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| 10:52.02 | wyoung | gusto: nice work |
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| 10:58.07 | gusto | no idea |
| 10:59.00 | gusto | I am bouncing here and there with my theories what could be wrong, but I do not believe it is openbsd's fault any more, OK, I set the buffer back to normal, now it runs OK |
| 10:59.29 | gusto | but when he is forwarding packets, the load increases ... until it hits 100% of cpu (user) and then restarts itself |
| 11:01.43 | gusto | and I also noticed that compared to other applications asterisk is running painfully slow, even with not many modules loaded, when I let him load all the modules, he hits the maximum at start and keeps himself in a restarting loop |
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| 11:07.43 | gusto | I am slowly starting to believe that there is a general problem with running asterisk on mips platforms, because I also experienced such behaviours, like when asterisk was just transcoding from 722 to 711 and back and it did fail miserably and that on openwrt, of course on openbsd thing are even worse, not to mention that this is mips64 |
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| 11:17.26 | wyoung | gusto: openbsd is great |
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| 11:17.32 | wyoung | same with freebsd |
| 11:17.41 | wyoung | and macosx (same thing) |
| 11:17.49 | wyoung | (darwin) |
| 11:18.09 | wyoung | gusto: what is the CPU temp? |
| 11:20.39 | gusto | I do not think that the cpu temperature is here rising, the edgerouter is not something that could get a lot of power from the PoE I have it on |
| 11:20.59 | gusto | I did put my hand on it and I find the temperature normal |
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| 11:22.06 | gusto | hm, it ends with a segmentation fault |
| 11:22.24 | gusto | to me it looks like some problem with the glbc |
| 11:22.55 | gusto | glibc, because openwrt had similar issues, but I have no idea how to find out with what lib it was compiled, or even if glibc is a part of asterisk bin |
| 11:24.16 | wyoung | Here is a quote - "Come on driver go!!! If I wanted to sit down for a living and not go anywhere I would become a teacher!" |
| 11:28.19 | gusto | lol, after releasing the asterisk rtp module, the cpu dropped to 0% load 100 % idle |
| 11:28.38 | gusto | and asterisk is not consuming any considerable resources any more, so it was somewhere on the rtp engine |
| 11:28.41 | gusto | lol |
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| 11:59.18 | wyoung | hi |
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| 12:01.34 | andycol_500 | Hi Guys |
| 12:02.00 | andycol_500 | im having a small problem when my customer receives a call and i transfer it, the callerid showing is not the outside number but instead the extension |
| 12:02.15 | andycol_500 | is there a way to set callerid to something like CALLERID(name)=asreceived) |
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| 12:34.27 | linux4life | andycol_500, set(CALLERID(name)=${CALLERID(name)}) |
| 12:34.56 | linux4life | andycol_500, just read your original question |
| 12:35.20 | andycol_500 | ok |
| 12:35.23 | andycol_500 | let me test |
| 12:35.27 | linux4life | if they are doing an attended transfer the callerid will show the extension number of the phone that transferred the call. |
| 12:35.45 | linux4life | if they are doing a blind transfer then it should show the callerid of the call being transferred. |
| 12:37.08 | linux4life | sounds like they are doing an attended transfer, which in that case the caller id behavior is as designed. |
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| 13:00.38 | andycol_500 | is there a way to show the callerid of outside during an attended? |
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| 13:23.36 | linux4life | andycol_500, the only thing I can think of off the top of my head is a dialplan entry, like a feature code. |
| 13:23.51 | linux4life | if you need help building it, pm me. |
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| 13:26.53 | Ast001 | Hello, is it possible in Asterisk 1.8 version to make MeetMe conference so that one of two conference users can call real number while in conference and to make that called client enter in same conference when he answers the call ? |
| 13:29.29 | Ast001 | or to make it simple I want to make a conference and then to call some user to join it by dialing his real phone number. |
| 13:31.31 | linux4life | only way I can think of is by calling that party on a separate session, and transfer them to the conference. |
| 13:31.53 | linux4life | short of that, I'm thinking you'd have to write your own confbridge app and hook it to ast by ari |
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| 13:37.40 | Samot | Asterisk 1.8 is dead. |
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| 13:40.04 | [TK]D-Fender | Leave the conference, Originate those outbound calls and rejoin the conference. |
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| 13:44.04 | Ast001 | hmm ok I will try that way thanks |
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| 13:51.58 | todwetsprock | join #openvpn |
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| 13:57.42 | linux4life | todwetsprock - forgot the leading / |
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| 14:23.12 | utrack | Hey! I've got two phones - testphoneA and testphoneB (http://pastebin.com/fGCqm2Pg) |
| 14:24.33 | utrack | One is udp softphone and another is in-browser softphone. I try to call to some external number from udp and it works both ways, but websocket one does not receive audio data |
| 14:25.22 | AviiNL | any messages from asterisk? |
| 14:25.23 | utrack | However, external phone can hear the websocket's microphone. Could anyone point me where to dig? |
| 14:25.28 | utrack | AviiNL: no errors |
| 14:25.48 | AviiNL | do you have headphones plugged in ? |
| 14:25.48 | utrack | AviiNL: calling works, one-way tho |
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| 14:26.11 | utrack | AviiNL: yup, sure |
| 14:28.37 | AviiNL | in rtp.conf is ICE configured? |
| 14:29.28 | AviiNL | and or stunaddr? |
| 14:29.33 | AviiNL | try adding stunaddr=stun.l.google.com:19302 |
| 14:30.38 | AviiNL | just noticed you do icesupport=yes on peer level instead of global (rtp.conf) which should be fine, try adding the previously mentioned line in your [webrtc] block |
| 14:31.20 | utrack | AviiNL: icesupport=true, stunaddr is set |
| 14:31.28 | utrack | AviiNL: oh, it should be yes instead of true? |
| 14:31.30 | utrack | typo |
| 14:35.10 | gusto | no problem with asterisk 1.8 being dead |
| 14:35.19 | gusto | but ... it was sofar one of the best releases |
| 14:35.51 | scv | 12800R DDR3L |
| 14:35.53 | scv | er |
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| 14:37.54 | utrack | nope, fixing icesupport=true globally doesn't work... as well as adding stunaddr |
| 14:39.58 | utrack | double-checked the hardware, calling via udp softphone works for the same box |
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| 15:19.42 | cusco | hi |
| 15:19.53 | cusco | is there a way I can make a sip peer not available |
| 15:20.03 | cusco | via cli.. ? |
| 15:21.34 | [TK]D-Fender | Firswall them out. |
| 15:21.38 | [TK]D-Fender | Firewall* |
| 15:22.59 | cusco | hmm.. |
| 15:23.35 | cusco | now that you mention that, I could change the host to 127.0.0.1 or so.. as this is via a sql backend |
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| 16:39.26 | sh0wbiZ | hu |
| 16:39.30 | sh0wbiZ | hi |
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| 16:39.58 | sh0wbiZ | i need some assitance in selecting the right IP PBX system |
| 16:40.02 | wyoung | hi |
| 16:40.03 | sh0wbiZ | anyone here willing to help me? |
| 16:40.16 | wyoung | I have 5 minutes, shoot |
| 16:40.20 | wyoung | what is your question? |
| 16:40.23 | wyoung | questions |
| 16:40.26 | sh0wbiZ | well a lot of questions |
| 16:40.39 | sh0wbiZ | even dunno how to start |
| 16:40.52 | wyoung | start at the beginging :) |
| 16:40.56 | sh0wbiZ | the company i'm working for wants to equip a school with an IP PBX system (school in in Sri Lanka) |
| 16:41.05 | sh0wbiZ | we are working on an international tender |
| 16:41.19 | wyoung | nice, is it a selective cricket school? |
| 16:41.23 | wyoung | ok |
| 16:41.27 | sh0wbiZ | they want PCs, servers etc but also an IP PBX system and phones |
| 16:41.30 | sh0wbiZ | haha no |
| 16:41.34 | wyoung | aawww |
| 16:42.06 | wyoung | well I would say ubuntu ubuntu all the way :) but that is just me, I don't know what apps you need to support |
| 16:42.06 | sh0wbiZ | i have a quote from a company in sri lanka for a pbx from a company called asttecs |
| 16:42.28 | sh0wbiZ | their system uses asterisk |
| 16:42.32 | wyoung | ok |
| 16:42.50 | sh0wbiZ | but i'm too unqualified to judge whether what they offered me complies with what is specified in the tender |
| 16:42.58 | wyoung | asterisk is solid, out of the box it isn't that user friendly, but that is fine if you are skilled enough to use it |
| 16:43.25 | sh0wbiZ | well i guess to company in sri lanka is skilled enough to do the initial installation and setup |
| 16:43.38 | wyoung | and it is usually a case of you configure it then leave it, it doesn't need any maintenence |
| 16:43.48 | sh0wbiZ | thats good |
| 16:44.20 | sh0wbiZ | is asterix an open standards communication server software |
| 16:44.26 | sh0wbiZ | asterisk of course :P |
| 16:45.09 | wyoung | It is open source, and it complies with open standards (SIP for example) |
| 16:45.26 | wyoung | SIP being the main protocol used in VoIP systems |
| 16:45.33 | sh0wbiZ | yes i thought so |
| 16:45.45 | wyoung | Asterisk also supports the superior IAX protocol, which is native to asterisk |
| 16:46.18 | WIMPy | As if SIP ws anything like a standard. |
| 16:47.16 | WIMPy | And Is somewhat dismissed in here. :-( |
| 16:57.30 | wyoung | WIMPy: quite you! It is used everywhere, it is a standard! |
| 16:57.46 | wyoung | I prefer IAX but most voip providers don't offer that |
| 16:57.50 | wyoung | quiet* |
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| 16:58.08 | wyoung | mrmccrac@@@@!!!!!!!!! |
| 16:58.29 | mrmccrac | er hello |
| 17:01.30 | wyoung | how you been buddy? |
| 17:01.34 | linux4life | Do you think there is really a market for IAX connectivity to providers? |
| 17:01.59 | linux4life | I would think if there was, we'd see more of them.... |
| 17:02.11 | wyoung | linux4life: that is why SIP is the standard |
| 17:03.42 | mrmccrac | im okay but dont understand this error "[2016-04-13 16:55:03] DEBUG[19712] pbx_spool.c: Delaying retry since we're currently running â¦" because the call file shouldn't still be running |
| 17:05.09 | mrmccrac | it successfully makes a call, asterisk removes it from the outgoing directory/ and then next time i try to do another call it claims the call file is currently running still |
| 17:05.25 | wyoung | ok |
| 17:06.07 | mrmccrac | ive since set the MaxRetries to 5 in the call file but it shouldn't need to do that.. |
| 17:06.20 | wyoung | ok |
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| 17:15.27 | marceloamorim | hello guys, I'll post my while code and could you help me find the problem inside the while? |
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| 17:20.23 | linux4life | post it via pastebin, and someone may help you.... post it here, and someone may hurt you.... ;-) lol |
| 17:23.22 | marceloamorim | http://pastebin.com/HxwEC6j6 |
| 17:25.06 | marceloamorim | well, te problem seems after the return, it hits the bitvoix:22, then after :24 back to 18 but after that straight 25 |
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| 17:25.57 | marceloamorim | at the line 18, the while sip/10.100.0.2, "0" this "0" should be 2 right? |
| 17:26.41 | marceloamorim | I used the verbose to check if the counter was wrong, but show me 2 and 2 is the right number |
| 17:36.37 | marceloamorim | expired |
| 17:36.39 | marceloamorim | the new one |
| 17:36.39 | marceloamorim | http://pastebin.com/WeLrQ8nk |
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| 17:41.28 | marceloamorim | if you see, the last one "while("SIP/10.100.0.2-00000333", "0") in new stack why is showing 0 =( |
| 17:54.09 | marceloamorim | yeah, I couldn't find the problem =( |
| 17:57.07 | marceloamorim | actually, fix the problem =) |
| 17:59.15 | RadicalDev | with pjsip, how can I add headers to the INVITE sent by a later DIAL command from an agi script? I can see that this call does kind of what I would expect, agi.set_variable("PJSIP_HEADER(add,X-Number)", "8885954952"), but the outgoing INVITE does not contain the header. |
| 18:01.38 | RadicalDev | Example from AGI debug: <PJSIP/inbound-00000008>AGI Rx << SET VARIABLE "PJSIP_HEADER(add,X-Number)" "8885954952" |
| 18:08.45 | file | PJSIP_HEADER has to be set on the outgoing channel, it is used in pre-dial handlers |
| 18:09.09 | file | https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers |
| 18:09.48 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 18:12.39 | RadicalDev | Fancy. Thanks, file |
| 18:13.49 | marceloamorim | http://pastebin.com/WeLrQ8nk the last while send "0" I think it should send "2" |
| 18:14.42 | [TK]D-Fender | Show us the actual call |
| 18:15.27 | marceloamorim | I show the full CLI from the call |
| 18:15.53 | [TK]D-Fender | ${ODBCROWS} |
| 18:16.07 | [TK]D-Fender | You should be NoOping this so you can prove how many it should loop for |
| 18:17.29 | marceloamorim | I use verbose to get the variable count, I'll try the odbcrows again then, nice call |
| 18:18.01 | [TK]D-Fender | Prove what you're STARTING with. Assumptions = bad |
| 18:20.24 | marceloamorim | yeah [TK]D-Fender . you are right, the second time I call the odbcrows it returns only 1 not 3 anymore |
| 18:20.37 | marceloamorim | I'll set a global variable for that and probably fix the problem |
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| 18:21.01 | [TK]D-Fender | You're supposedly trying to loop through ODBC results... you shouldn't be assuming how many records either... |
| 18:21.55 | marceloamorim | it shows the number before, I assume it keeps the same number, but I'm wrong the way I think about how it works |
| 18:23.41 | rekrul | so, I am seeing strange behavior in asterisk. I am ACKing a 200 OK, logs say that set_destination is getting set to the IP I expect but my packet capture is going out to a different IP. |
| 18:24.36 | rekrul | set_destination: set destination to 10.166.226.38:5060, it's actually going to a different IP and I can;t figure out where it might be getting changed |
| 18:25.54 | rekrul | Transmitting (no NAT) to 10.166.226.38:5060: (this is the IP I expect but actually going to a SIP router in the network) so the ACK isn;t getting to the right place and the network wont start the audio stream |
| 18:30.07 | rekrul | when I had outboundproxy set, would set_destination to the wrong IP (SIP Router IP) in teh logs. After removing it it now says teh write IPs in the logs but sending the wrong IP on the wire. |
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| 18:46.17 | marceloamorim | [TK]D-Fender: ty so much dude to light me |
| 18:46.43 | marceloamorim | could fix the odbc "problem" |
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| 19:20.44 | RadicalDev | with PJSIP, I'm trying to figure out why asterisk is setting the contact header to "Contact: <sip:asterisk@1.2.3.4:5060>". I think one of the carriers I'm dailing over is using the contact username as the caller ID which doesn't work well when that value is 'asterisk' |
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| 22:59.22 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.0 (2016/03/29), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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