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| 15:43.50 | martinvw | We've set up an IVR for our hotline that looks something like this: https://pastebin.com/fUvsZC8u |
| 15:43.50 | martinvw | The customer dials our number (e.g. 123456), and can then press 1 to hear a canned announcement, or he can press 2 to be patched through to our support hotline. This works fine, but has an unexpected site effect: when a customer that has pressed 2 uses the automatic redial function on his phone, the phone calls 1234562, and he ends up on a different extension. |
| 15:44.54 | martinvw | Is there any way to prevent this? Can I somehow tell the phone that whatever is entered in WaitExten() is not part of the phone number and should never be appended when using the automatic redial function? |
| 15:46.01 | Samot | You can't. |
| 15:46.13 | Samot | *their* phone is storing all the digits. |
| 15:46.34 | Samot | When they hit redial on *their* phone it redials what is in the history |
| 15:50.19 | martinvw | Samot: I had expected as much, but had hoped that there's some kind of DTMF magic that asterisk might send to the phone. oh well, but thanks for the reply! |
| 15:51.18 | martinvw | I haven't even been able to reproduce the issue; my iPhone doesn't seem to have this behaviour (the call history will only show the original number without the input appended to it), and it also has no automatic redial feature |
| 15:52.24 | martinvw | but I have gotten several complaints from the staff that handles the different extension where the people accidentally end up, so the problem does seem to exist |
| 15:52.34 | igcewieling | I can't even imagine how that would happen. |
| 15:53.16 | igcewieling | a SIP phone should send the entire string, causing extension not found issue. A PSTN call won't pass the extra digits. What do you have, analog phones on a DAHDI port? |
| 15:53.50 | martinvw | we're using SIP softphones internally. no idea what kind of phones the affected customers use |
| 15:54.18 | martinvw | the problem is, the other extension *does* exist |
| 15:55.02 | martinvw | so if our support hotline is 123456-1, and the customer presses 2, his phone stores 123456-12 as the dialled number, and -12 is actually an extensions in our system |
| 15:56.30 | martinvw | if we had known about this behaviour before, we'd probably have used something like 0001 as the support hotline |
| 15:57.08 | igcewieling | well, stop doing it that way |
| 15:58.53 | igcewieling | and, no I meant don't have an extension 12345612 |
| 15:59.49 | igcewieling | perhaps if you pastebinned the CLI output of one of these problem calls someone can provide useful advice. |
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| 16:07.44 | martinvw | igcewieling: not sure how to do this. the complete call log contains a lot of private information like employee names; I would have to mangle it pretty thoroughly before I could post it, and I'm not sure how useful it would be afterwards |
| 16:10.23 | tmoore | hey asterisk team hope y'all are doing well! I'm having an issue with passing caller-id through when dialing a pjsip contact. From the Asterisk 18 documentation I should be able to use the dial options "o" or "f" to force the caller-id from one leg to the next, but I can't seem to get this working... here is an example from dialplan: |
| 16:10.30 | tmoore | [tracked-dial] |
| 16:10.31 | tmoore | ... |
| 16:10.36 | tmoore | exten => s,n,Dial(${ARG1},${ARG2},o(${CALLERID(all)})${ARG3},${ARG4}) |
| 16:10.39 | tmoore | ... |
| 16:11.48 | martinvw | this is the beginning of the log, including a DumpChan() output: https://pastebin.com/RgF5g1UH |
| 16:11.53 | igcewieling | martinvw: I can't imagine how that would be the case, but whatever. |
| 16:12.23 | igcewieling | martinvw: I don't see the Dial line. |
| 16:12.25 | tmoore | My expectation is that the "From" header will now contain the caller-id info from A-leg caller when dialing B-leg callee, but Asterisk does not set it... |
| 16:12.43 | igcewieling | ....er tmoore: I don't see the Dial line. |
| 16:12.57 | martinvw | the last line shows the problem: the customer had intended to dial xxxxxxxxx01, but dialled xxxxxxxxx012 instead |
| 16:13.11 | tmoore | exten => s,n,Dial(${ARG1},${ARG2},o(${CALLERID(all)})${ARG3},${ARG4}) |
| 16:13.30 | igcewieling | no, that is not the line as shown in the CLI with all the variables replaces with thevalies we need to see. |
| 16:13.53 | tmoore | oh gotya: Executing [s@tracked-dial:6] Dial("PJSIP/gate-02.voipmuch.com-00000000", "PJSIP/1000/sip:1000@reg-01.voipmuch.com:5060,15,o("Tyler Moore" <12489092769>),") |
| 16:14.56 | igcewieling | OK. That looks good. You might consider removing the quotes, but the format is vaild. Is the callerid value you set a valid DID on your account? |
| 16:16.49 | tmoore | igcewieling: it's going to one of our proxies so it will basically accept it no matter what the "From" is, as long as it is valid, and route from there. |
| 16:16.49 | igcewieling | martinvw: my point is that dialing xxxxxxxxx012 is matching something and it should not be. |
| 16:19.01 | igcewieling | martinvw: your PBX should be rejecting calls to xxxxxxxxx012 |
| 16:20.19 | tmoore | iqcewieling: I also tried passing the caller-id using ${CONNECTEDLINE(...)}, and tried setting "trust_id_inbound=yes" and "trust_id_outbound=yes" with no luck... |
| 16:20.51 | igcewieling | I always set that stuff in my pre-dial handler. |
| 16:21.42 | Samot | Connected line has nothing do to with sending CallerID out. |
| 16:21.48 | igcewieling | CALLERID should be passed to the outgoing channel when set on the incoming channel, but it has been a long time since I used it that way. |
| 16:22.43 | tmoore | Samot: i was getting desperate... lol, the documented Dial() options should work but don't seem to be in my environment |
| 16:23.50 | martinvw | igcewieling: but xxxxxxxxx012 is a valid extension. We have a phone block with 1000 numbers, and have structured it like this: 0 central office, 01 support, 02 sales, and so on. 01x are specialized support extensions, 02x would be specialized sales extensions and so on |
| 16:24.38 | Samot | martinvw: Are these outside callers or internal callers? |
| 16:24.47 | martinvw | igcewieling: if your point is "don't use 01x extensions when you're running an IVR on 01", then well yes, we realized this now :D |
| 16:24.53 | martinvw | Samot: external callers |
| 16:24.56 | igcewieling | Samot is talking now, you'll have to deal with him. |
| 16:26.37 | martinvw | In retrospect, we were probably stupid when structuring our phone numbers, but back then we neither had planned to use IVRs, and we didn't know about this problem |
| 16:27.26 | martinvw | As a workaround, we have now moved the 012 extension to 019 where it doesn't conflict with the IVR |
| 16:28.25 | martinvw | I had just hoped that there might be a cleaner solution to completely prevent this conflict that I don't know about but someone here would :) |
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| 16:36.51 | igcewieling | are you in germany? |
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| 16:52.27 | EmleyMoor | How do I create a subroutine that takes a delimited list of parameters as one of its arguments, and repeats for each of them? |
| 16:52.39 | tmoore | iqcewieling: do you have an example you used from your pre-dial handler? maybe that could help me further troubleshoot why my version isn't working |
| 16:57.16 | Samot | EmleyMoor: You use a While statement |
| 16:57.49 | igcewieling | same = n,Set(PJSIP_HEADER(add,P-Asserted-Identity)="${MASTER_CHANNEL(sm_call[${call_index}][caller_tnis_name])}" <sip:${MASTER_CHANNEL(sm_call[${call_index}][ani])}@${SM_NODE_IP}>) |
| 16:57.49 | igcewieling | <PROTECTED> |
| 16:58.26 | igcewieling | I use PAI, which would normally set the callerid for me. |
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| 17:00.57 | EmleyMoor | Samot: Well, that much is no surprise... |
| 17:01.16 | Samot | OK so what more do you need? |
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| 17:02.49 | EmleyMoor | Presumably the CUT function returns just the contents of its varname parameter as field 1 if the delimiter is not in it |
| 17:02.51 | EmleyMoor | ? |
| 17:03.27 | tmoore | igcewieling: hmmm maybe I can get around this by using rpid or pai.. i'll try that instead |
| 17:03.52 | EmleyMoor | (therefore an empty as field 2?) |
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| 17:04.44 | Samot | OK |
| 17:04.59 | Samot | I'll ask a different way, what are you looking for? |
| 17:05.00 | igcewieling | this is my dial line, but I don't think it will help. same = n,Dial(${cur_dest},${MASTER_CHANNEL(sm_call[${call_index}][dial_timeout])},gb(pre_dial^${MASTER_CHANNEL(sm_call[${call_index}][dnis_orig])}^1)U(answer)); |
| 17:05.10 | Samot | Because apparently you know a While and CUT is needed. |
| 17:06.57 | EmleyMoor | Samot: Mainly information on what CUT does (a) in the absence of a delimiter (b) with an out of range fieldspec.The docs I've got explain what it does except in those circumstances |
| 17:08.42 | tmoore | igcewieling: set send_rpid='yes',send_pai='yes' and "From" was also changed... that doesn't make sense to me but whatever it seems to work |
| 17:11.22 | EmleyMoor | I've only used CUT when I've known there would be something to cut before, this time there only might be |
| 17:12.09 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_CUT |
| 17:12.50 | EmleyMoor | Yes, but my what-ifs are still what-ifs. |
| 17:13.05 | Samot | Have you tested it? |
| 17:13.31 | Samot | Are you concerned there won't be something to CUT? |
| 17:13.37 | Samot | Or there only one entry? |
| 17:14.06 | EmleyMoor | I'm mainly concerned with how exhaustion of entries would be detected |
| 17:14.17 | Samot | You mean the last one? |
| 17:14.22 | EmleyMoor | I will cond= |
| 17:14.32 | EmleyMoor | last-or-only |
| 17:14.32 | Samot | If there is no delimiter after the last one it's done |
| 17:14.44 | Samot | One entry would loop once |
| 17:14.49 | Samot | 5 entries would loop 5 times |
| 17:15.26 | EmleyMoor | So, field 2 with no delimiter in the string would return an empty, ditto field 3 if there's only one delimiter? |
| 17:17.28 | EmleyMoor | I will put some testing in as I deploy but I am doing a comprehensive rewrite of part of my dialplan which is currently a mixed design based on the needs of initially two users and later one, due to there being another second user forthcoming |
| 17:18.47 | tmoore | thanks for your help everyone, your the best! |
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| 21:14.38 | mltngpot | can anyone help me compile Asterisk 1.8.32.3? I inherited have been having problems migrating from v1.8 w/sip => v18 w/pjsip, and I figure the best way for me to learn and to successfully do this is one version at a time. unfortunately I have not gotten it to build successfully yet |
| 21:15.37 | mltngpot | it craps out with the linker when trying to make chan_iax2.so |
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| 21:23.44 | seanbright | mltngpot: pastebin the linker error |
| 21:23.45 | seanbright | ~pb |
| 21:23.45 | infobot | 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: https://pastebin.com, https://paste.ubuntu.com, http://paste.debian.net; or install pastebinit with yum or aptitude. |
| 21:24.31 | seanbright | kbye |
| 21:24.45 | seanbright | mpot: pastebin the linker error |
| 21:26.01 | mpot | https://pastebin.com/UR0rhGR9 |
| 21:26.47 | seanbright | good god |
| 21:26.56 | mltngpot | thanks |
| 21:27.15 | mltngpot | that sounds positive |
| 21:27.28 | seanbright | this is just vanilla 1.8.32.3? |
| 21:27.33 | seanbright | no custom patches or anything? |
| 21:28.12 | mltngpot | just the get mp3, but format_mp3 is not selected |
| 21:28.37 | seanbright | i don't know what that means |
| 21:29.24 | mltngpot | contrib/scripts/get_mp3_source |
| 21:29.27 | seanbright | got it |
| 21:32.36 | seanbright | yeesh |
| 21:33.10 | mltngpot | cd .. |
| 21:36.00 | mltngpot | I just deleted the directory and untarred just to be sure i have a clean source tree |
| 21:36.33 | seanbright | my guess is that at the time it was released the linker was more forgiving |
| 21:37.33 | mltngpot | so perhaps trying on an older os |
| 21:37.35 | seanbright | honestly you are going to have better luck just skipping forward to 13 or something |
| 21:37.39 | seanbright | or trying on an older OS, sure |
| 21:37.51 | seanbright | i mean it's going to suck either way |
| 21:38.04 | seanbright | you can move to 18 with chan_sip |
| 21:38.10 | seanbright | then worry about chan_sip -> chan_pjsip |
| 21:57.07 | mltngpot | Thanks, I appreciate the help |
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