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01:02.29 | funxion | anyone here |
01:27.51 | Samot | There isn't "hunting" |
01:28.10 | Samot | They are going to present a list of codecs in the order they want to use them, individually.. |
01:28.19 | Samot | They go down the list until there's a match. |
01:29.57 | Samot | So it would depend on the codecs and their order you are sending. |
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02:34.06 | Maver|cK | Can i Make the Caller id of number which show in my 1st line of dialplan ? |
02:34.20 | Maver|cK | Executing [1122222222@from-goip:1] NoOp("IAX2/1234-9783", "$EXTEN") in new stack |
02:34.33 | Maver|cK | For examplle i want to make 11222222 to my caller id |
02:34.37 | Maver|cK | is it possible |
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02:38.36 | Maver|cK | ? |
02:38.46 | Maver|cK | any one alive ? to answer my question |
02:42.38 | Samot | ${CALLERID(num)} |
02:52.56 | Maver|cK | okay let me check |
02:54.01 | Maver|cK | Set(CALLERID(dnid)=${CALLERID(num)} |
02:54.05 | Maver|cK | like this ? |
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02:57.50 | WIMPy | Are you still in to this thing and still confusing everyone including yourself with your phrasing? |
02:58.14 | WIMPy | Or is it really something new? |
02:58.47 | Maver|cK | WIMPy bro em Newbie to Asterisk Em Stuck here from last 3 days :( |
02:59.31 | WIMPy | And I told you two ways to fix your issue yesterday. |
03:00.16 | Maver|cK | can u tell me |
03:00.24 | Maver|cK | Executing [1122222222@from-goip:1] NoOp("IAX2/1234-9783", |
03:00.31 | Maver|cK | this 1122222 i want ot make caller id |
03:00.41 | WIMPy | Either you save the extension in some other variable before it changes or you make sure it doesn't change by using Read() instead if WaitExten(). |
03:01.46 | WIMPy | And then you use your variable or EXTEN. Forget about CALLERID(dnid). |
03:04.38 | Samot | Wow.. |
03:04.40 | Samot | No.. |
03:04.47 | Maver|cK | WIMPy in which variable i can save this extensions ? |
03:04.59 | Samot | Set(CALLERID(num)=${EXTEN}) |
03:04.59 | WIMPy | You can name it whatever you want. |
03:05.15 | Maver|cK | okay |
03:05.20 | Maver|cK | let me try it |
03:05.34 | Samot | I think this like the 6th time someone has said "variable" |
03:06.46 | WIMPy | And it's not about caller ID at all, despite dnid is an option of the CALLERID function. |
03:08.36 | WIMPy | And using dnid was not a good idea anyway. 1. it's contents may differ depending on tech and 2. it might not contain what you expect. |
03:08.46 | Maver|cK | u Guys Are AWesome (Y) |
03:08.53 | Maver|cK | Thanks ALot I Got it What i Need (Y) |
03:09.03 | Maver|cK | GOd BlesS ALL of U (Y) |
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03:26.19 | lvlinux | Samot: nope, when I reloaded features.conf the console said that it was invalid and had to be 'caller', 'callee', or 'both' |
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05:36.32 | igcewieling | Samot: Constants aren't. Variables don't. -- unknown |
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06:39.17 | denysonique | Does ifTime even work in AEL? |
06:39.34 | denysonique | Do timelimited includes work? |
06:39.39 | denysonique | I tried, and they don't seem to... |
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09:34.57 | WildyLion | Hi guys. I need a bit of advice on queue logic |
09:36.11 | WildyLion | How do I write a dialplan (AEL) rule for a case when the caller hung up waiting in a queue vs just going to 'h' extension unconditionally? |
09:36.27 | WildyLion | even after the queued call has been completed successfully |
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13:06.52 | Tiffon | nas |
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15:29.21 | funxion | Anyone here? |
15:31.39 | [TK]D-Fender | nope |
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15:58.58 | _boot | hi, i've built asterisk 13 on a debian box, i'm trying to set up pjsip and the only thing i've been unable to do so far is get a transport with protocol=tls recognised - i see no errors in asterisk's startup log but the transport isn't there (it works fine if I use udp). libssl-dev is installed, what could i check? |
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16:29.46 | funxion | whats the most common reason a sip call dies after 8 seconds ? |
16:30.27 | eric_hill | Does anyone have recommendations on an IP-enabled FAX machine? Something that does iFax so I can integrate with Asterisk fax send/receive? |
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16:47.50 | igcewieling | funxion: direct media enabled when one end is behind NAT |
16:48.06 | igcewieling | iFax? |
16:48.12 | funxion | yup |
16:48.30 | igcewieling | is that a protocol or a product? |
16:49.13 | igcewieling | funxion: set directmedia=no and see if that helps. |
16:50.13 | funxion | that wasnt the problem |
16:50.48 | igcewieling | I did answer your question though. |
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16:51.43 | funxion | I was uplate troublshooting a carrier issue and tried adding their subnet as a localnet in sip.conf and forgot to remove |
16:51.53 | funxion | you did answer my question though yes |
16:51.55 | funxion | thanks |
16:52.20 | funxion | Im going nutz trying to figure out this issue with one of my carriers |
16:52.25 | igcewieling | funxion: misconfigured asterisk would also cause that symptom. 8-) |
16:52.26 | funxion | trying everything |
16:52.32 | funxion | correct |
16:53.08 | funxion | Maybe you can help me with it |
16:53.49 | funxion | I havea DID provider where some DID's work just fine and others work every 15 calls |
16:54.09 | funxion | I have iptables rules allowing there entire subnet |
16:54.39 | funxion | the calls that fail I see invites in ngrep but not in sngrep |
16:55.01 | igcewieling | does disabling iptables fix the issue? |
16:55.08 | funxion | nope |
16:55.31 | igcewieling | carriers, such as Verizon SIP use many different devices outside their SIP signaling IPs. |
16:55.43 | funxion | neither does allowing anonymous sip |
16:56.11 | funxion | I know all of the media gateways for the provider fall into the subnets allowed in iptables |
16:58.05 | igcewieling | Nice to have a provider like that. Most only tell customers where the SIP proxies are, not the RTP devices. |
17:13.56 | eric_hill | igcewieling: iFax means the machine sends outbound faxes as email to 5551212@somedomain.com and checks an IMAP box for inbound faxes. |
17:14.39 | eric_hill | igcewieling: I set up dovecot and postfix with asterisk to handle it, but I'm looking for a smaller device. The large copiers do it fine. I need a desktop version. |
17:18.27 | igcewieling | eric_hill: that has nothing to do with Asterisk. |
17:18.59 | igcewieling | You could build something with Asterisk if you really wanted to, I guess. |
17:34.18 | funxion | why would you be able to see an invite in ngrep but in sngrep? |
17:34.56 | Samot | I'm not sure that question is phrased properly. |
17:37.10 | igcewieling | funxion: I can think of no reason other than you are using them wrong. |
17:37.15 | funxion | Im troublshooting an issue where certain DID's work from from a specific carrier and others do not consistantly. The calls follow the same paths. The only thing I have noticed is that on the DID's that fail I can only see the invites in ngrep but not in sngrep. The calls that terminate properly the invites show up in sngrep at the same time as ngrep |
17:37.46 | funxion | igcewieling: I don't think I am using them wrong |
17:38.00 | funxion | I'm using sngrep filtering only for invites |
17:38.09 | igcewieling | ah, I suppose if the packet is not valid SIP, sngrep would fail. |
17:38.28 | funxion | ngrep -pqtN 'carrier subnet here' -W byline -d any port 5060 |
17:38.53 | igcewieling | I normally use tcpdump. |
17:38.59 | funxion | igcewieling: thats what I think but carrier is arguing |
17:39.08 | funxion | I see the same in tcpdump |
17:39.54 | igcewieling | don't filter port 5060, filter on source IPs if the carriers. |
17:40.00 | igcewieling | s/if/from |
17:40.59 | Samot | Are you using Chan_SIP or Chan_PJSIP? |
17:41.22 | funxion | chan_sip |
17:41.47 | Samot | Is "sip set debug on" not giving you the information you need? |
17:42.06 | Samot | Or even "sip set debug ip <peer ip>"? |
17:42.10 | funxion | on the calls that fail there is no output in sip debug |
17:42.18 | funxion | I only see the invites in ngrep |
17:42.21 | funxion | or tcpdump |
17:42.50 | Samot | And are they being sent to the right IP and port that Chan_SIP/ASterisk is listening on? |
17:43.17 | igcewieling | how about posting ngrep if the failed call? someone might be interesting slogging through the info. |
17:43.22 | funxion | yup |
17:43.32 | funxion | sure |
17:43.36 | Samot | If Apache isn't listening on 8080, ngrep will show the traffic but the apache log will not. |
17:43.39 | funxion | give me a few to gather the pieces |
17:44.38 | igcewieling | except for the most obvious errors, I don't troubleshoot packet captures, but there are others on this channel who might. |
17:50.07 | funxion | http://pastebin.com/LRv3G4Mp |
17:50.21 | funxion | All there is is inbound invites without responses |
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17:53.02 | Samot | Show a "working" one. |
17:56.39 | funxion | I love the quotes around working BTW |
17:57.45 | igcewieling | heh "User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150416T155219Z~6e247a3f3d~64bit." |
17:59.00 | igcewieling | try setting nat=yes. U(17) 2016/12/29 17:45:40.899262 8.27.107.6:5060 -> 10.100.253.4:5060 looks like a NAT issue, depending on the interface |
17:59.03 | Samot | Is it "coming"? |
17:59.15 | funxion | inbound yes |
18:00.09 | igcewieling | what codecs do you have enabled? try enabling only ulaw (or alaw if you are outside north america) |
18:00.49 | Samot | Codecs or NAT issues would still show up in a sip debug in the CLI |
18:00.51 | funxion | I only have ulaw enabled |
18:00.59 | funxion | I beleive it to be a NAT issue |
18:02.35 | igcewieling | make sure you have nat=yes localne and externip set. |
18:04.55 | Samot | Where is a working DID call? |
18:06.41 | igcewieling | also set qualify=10000 and qualifyfreq=30 to see if that helps |
18:08.24 | Kaian | funxion, are you using sngrep with port 5060 filters? Your packages are fragmented by IP, so try capturing without bpf filters |
18:08.59 | igcewieling | (12:39:54 PM) igcewieling: don't filter port 5060, filter on source IPs if the carriers. |
18:10.50 | Samot | Why is all this being suggested?! |
18:11.09 | Samot | There is an issue with certain DIDs from the same provider on the same peer not working. |
18:11.27 | Samot | They show up in ngrep but not in an Asterisk console sip debug... |
18:11.58 | igcewieling | Hmm? NAT translations on routers sometimes clear the translation if no UDP traffic has been received. Causes inbound calls to randomly fail. That is why I suggested qualify and qualify freq. |
18:12.02 | Samot | So they are hitting the server but are they hitting an ip and port Asterisk is listening on? |
18:12.19 | Samot | I want to see the difference between a failed DID and a working DID |
18:12.35 | Samot | To see if they are being sent the same way with the same information. |
18:12.56 | Samot | Compare the two data sets before making a bunch of change suggestions. |
18:13.21 | igcewieling | have fun, I'm going back to useful stuff. |
18:13.29 | Samot | I'm about to do the same. |
18:13.53 | Samot | I've waited 20 minutes for requested debug information and it's not coming. |
18:15.27 | funxion | sorry carrier called |
18:16.14 | funxion | and I think they changed something because I cant get a "working" call now |
18:17.34 | funxion | got one |
18:17.37 | funxion | coming in a sec |
18:17.53 | funxion | http://pastebin.com/aXTSHasF |
18:21.25 | [TK]D-Fender | funxion, Stop showing external dumps and prove whateach channel driver is listening on, etc |
18:28.07 | Samot | That's the issue. The call doesn't show up in the Asterisk CLI with a sip debug. |
18:29.32 | Samot | funxion: OK, now do this: |
18:29.36 | Samot | asterisk -rvvvvv |
18:29.49 | Samot | sip set debug ip <peer ip> |
18:29.58 | Samot | Make a call on a DID that doesn't "work" |
18:30.20 | Samot | Copy and paste EVERYTHING from the moment you enter the Asterisk CLI to hanging up the call. |
18:30.21 | Samot | ~pb |
18:30.22 | infobot | from memory, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
18:30.29 | Samot | ^ Give us the link. |
18:33.05 | funxion | there is a lot of output its a live pbx |
18:33.25 | funxion | and the call that is failing does not show up in sip debug in asterisk |
18:33.37 | [TK]D-Fender | Go prove firewalls and SIP settings |
18:34.21 | funxion | bindport=5060 is in sip.conf |
18:36.13 | [TK]D-Fender | "sip show settings" |
18:36.19 | [TK]D-Fender | dump your firewall. |
18:36.37 | [TK]D-Fender | We need actual proof &dumps shown |
18:36.44 | [TK]D-Fender | anything less is a waste of everyone's time |
18:37.02 | funxion | k |
18:37.19 | funxion | http://pastebin.com/VZb6wAb8 sip settings |
18:39.05 | funxion | http://pastebin.com/xdvSjQCr iptables -L output |
18:42.22 | [TK]D-Fender | 10.100.253.4/255.255.255.255 |
18:42.33 | funxion | internal ip of asterisk box |
18:42.39 | [TK]D-Fender | Why do you have a host entry as NETWORK one? |
18:42.52 | [TK]D-Fender | You should not be making host entries like this |
18:43.01 | [TK]D-Fender | it's not loke * should be talking to itself |
18:43.11 | [TK]D-Fender | like* |
18:44.01 | funxion | does that really matter |
18:45.11 | funxion | removed |
18:47.00 | [TK]D-Fender | ACCEPT udp -- anywhere anywhere state RELATED,ESTABLISHED |
18:47.06 | [TK]D-Fender | UDP = stateless |
18:47.10 | [TK]D-Fender | so this looks funny. |
18:48.17 | funxion | what would you suggest |
18:49.13 | [TK]D-Fender | well first if you see see sip debug for other things, but not what that external dump showed, then it's pretty much hard proof that your firewall is responsible |
18:49.41 | [TK]D-Fender | So validate taht by temporarly flushing it entirely so you should expect to see your call attempt |
18:49.52 | [TK]D-Fender | Then time to reinstate and review it all |
18:50.43 | funxion | I have flushed it and get the same results |
18:51.03 | [TK]D-Fender | * has not filtering of its own. |
18:51.06 | [TK]D-Fender | This has to be external |
18:53.35 | funxion | Noramally I would agree with you on this but I have had iptables off with the same results |
18:53.49 | funxion | I can see invites in ngrep and tcpdump but not snmgrep or asterisk |
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18:53.57 | [TK]D-Fender | This particular point isn't really up for debate. |
18:53.59 | [TK]D-Fender | * has no firewall |
18:54.04 | [TK]D-Fender | So whatever you have is not part of * |
18:54.20 | [TK]D-Fender | So either the defaults or rules or bad or something else is in the way |
18:54.54 | [TK]D-Fender | Ther is no filter, nothing to "bug" no extra settings. * shows the packets when you enable SIP debug. Nothing shown = nothing made it to the daemon |
18:55.10 | funxion | agreed |
18:55.25 | funxion | my thoughts is that the packets are malformed |
18:55.40 | funxion | thats why they do not show up in sngrep either |
18:56.15 | funxion | but I can't see anything wrong with them |
18:56.22 | funxion | was hoping someone else would |
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20:06.11 | igcewieling | gads, I really hate Javascript. |
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21:13.26 | funxion | so I think I've zeroed in on the cause of my problem just not sure how to resolve it |
21:13.52 | funxion | the sip invite is getting fragmented and centos is timing out on reassembly |
21:14.02 | funxion | thus not answering the invite |
21:14.09 | funxion | or getting to asterisk |
21:14.16 | funxion | not an iptables issue |
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22:20.51 | WIMPy | [01:31] simplydrew has joined #asterisk (~simplydre@unaffiliated/simplydrew) |
22:20.54 | WIMPy | [01:40] tomcruise has left IRC (Ping timeout: 265 seconds) |
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22:21.06 | WIMPy | Argl |
22:21.12 | tomcruise | ? :) |
22:21.59 | WIMPy | I just hit the mouse again with clutter. |
22:22.03 | tomcruise | haha ok |
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22:26.43 | funxion | Samot: [TK]D-Fender |
22:26.52 | funxion | I figured out my issue and wanted to share |
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22:30.06 | funxion | The reason the daemon didn't see the invites is because the invites were being fragmented |
22:30.24 | funxion | freeswitch was adding too many information elements |
22:30.42 | funxion | once the stuffing was removed fromt he freeswitch side everything is working |
22:31.01 | WIMPy | ICE? |
22:31.38 | funxion | ? |
22:31.56 | WIMPy | Caused by ICE? |
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