| 00:00.02 | [TK]D-Fender | set it for your peer |
| 00:00.42 | [TK]D-Fender | This forces * upon matching them to ignore the remote RTP offer |
| 00:10.19 | sicelo | thanks. moving it to peer works much better |
| 02:26.15 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 02:26.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 14:27.33 | ganapathi | hi |
| 14:27.46 | ganapathi | anybody please clarify me |
| 14:27.59 | file | ask a question and someone may answer |
| 14:28.21 | ganapathi | am using SIP trunk for calling. in that time am unable to listen original network announcement. |
| 14:31.05 | [TK]D-Fender | what time? |
| 14:31.21 | [TK]D-Fender | What announcement? |
| 14:32.56 | ganapathi | ringing |
| 14:33.06 | ganapathi | user busy.switched off. |
| 14:33.15 | ganapathi | while dialing manual call. |
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| 14:33.57 | ganapathi | instead of original ringing look like some fake ringing sound playing to the user. |
| 14:35.04 | ganapathi | it's normal ?. or can able to cofigure tweaking to do solve this ?. or it may pbm with destination side ? |
| 14:36.07 | scampbell | Could the wrong country be set in the indications.conf |
| 14:37.22 | scampbell | ganapathi: I'm fairly novice with asterisk still but take a look at your indications.conf file. |
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| 14:38.15 | ganapathi | if i use DAHDI then working fine. |
| 14:38.20 | ganapathi | PRI |
| 14:38.48 | [TK]D-Fender | indications.conf is only used for inband signalling |
| 14:38.59 | [TK]D-Fender | so depends whoat yuor call is doing |
| 14:39.06 | [TK]D-Fender | Go look at the calls |
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| 14:41.15 | ganapathi | only ringing tone is changed. |
| 14:41.37 | ganapathi | but i didn't get original ring which provided by PRI network of the customer |
| 14:44.24 | [TK]D-Fender | Go look at the call. |
| 14:45.28 | [TK]D-Fender | normally you shouldn't get tone from them. PRI is digital signalling so they normally jus pass the state and your own system typically forwards that if this was started from another inbound call that hasn't been answered. |
| 14:45.49 | [TK]D-Fender | in that case its the phone you're listening on. If it has been answered before dialing out, then it's indications.conf |
| 14:49.40 | ganapathi | So it means you are saying, destination SIP user need to be answered to pass original voice data through SIP trunk ? |
| 14:51.49 | ganapathi | created peer sip in asterisk and connected to another telephony. when call made from asterisk then it's dialing SIP trunk which is in another telephony, but sip trunk on next telephony need to answer before dialing out to get original voice ?. |
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| 14:56.09 | [TK]D-Fender | yes |
| 14:56.17 | [TK]D-Fender | There is no "original voice" |
| 14:56.36 | [TK]D-Fender | and don't use the term "voice" unless you're referring to the called party actually answering and talking to you |
| 14:57.24 | ganapathi | ok got it. |
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| 15:03.43 | ganapathi | is there any configuration for auto answer the sip user ?. |
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| 15:11.00 | [TK]D-Fender | That's up to the endpoint |
| 15:11.15 | ganapathi | ok.if i use asterisk as endpoint then ? |
| 15:11.21 | ganapathi | what would be |
| 15:11.25 | [TK]D-Fender | You're not being clear |
| 15:11.40 | [TK]D-Fender | rephrase your entire question and be complete about each piece involved |
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| 15:12.54 | ganapathi | See. i have two telephony system. 1.asterisk. 2. yate. |
| 15:13.22 | ganapathi | i use yate for ISDN PRI based calling. |
| 15:13.51 | ganapathi | in that created sip user for asterisk to use as sip trunk. |
| 15:14.09 | ganapathi | am passing call from asterisk to yate via SIP trunk . |
| 15:14.36 | ganapathi | it's returned me as fake ringing until customer answered. |
| 15:15.22 | [TK]D-Fender | Draw a clearer line. |
| 15:15.35 | ganapathi | here my question is to original ringing tone which is provided by ISDN customer |
| 15:15.36 | [TK]D-Fender | A > B > C > D |
| 15:15.54 | [TK]D-Fender | And the tech involved |
| 15:16.09 | ganapathi | Asterisk -> SIP trunk -> Yate -> ISDN PRI. |
| 15:23.38 | [TK]D-Fender | How is Asterisk the start of this process? |
| 15:23.53 | [TK]D-Fender | Are you doing telemarketing call-outs from it? |
| 15:24.52 | ganapathi | no. i dialing manually as a person using asterisk |
| 15:25.44 | ganapathi | telecaller -> SIP User -> asterisk -> SIP trunk -> Yate - > ISDN PRI -> Customer. |
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| 15:26.45 | [TK]D-Fender | So far all of that is typically out-of-band. |
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| 15:27.05 | [TK]D-Fender | so the telecaller's phone is the one generating ring-tones, etc |
| 15:27.21 | [TK]D-Fender | unless one of those pieces in the the chain is forcing an answer. |
| 15:27.38 | [TK]D-Fender | at which point I've already recommended looking at the call to verify this |
| 15:28.34 | ganapathi | ok thanks. looking into autoanswer feature on asterisk and yate to do the same. |
| 15:28.58 | [TK]D-Fender | I don't think you're using that term correctly |
| 15:29.29 | [TK]D-Fender | * dialplan answers calls. Endpoints * is talking to can answer calls. |
| 15:29.36 | [TK]D-Fender | * doesn't "auto-answer". |
| 15:30.00 | [TK]D-Fender | It can ask that an endpoint it is calling to do so depending on what it supports. |
| 15:30.40 | ganapathi | in asterisk SIP user it's answered by dial plan |
| 15:31.19 | [TK]D-Fender | Still not clear |
| 15:31.31 | ganapathi | am looking into auto-answer on End point, means in YATE. |
| 15:31.48 | ganapathi | but at the same time. looking into reverse format. |
| 15:31.56 | [TK]D-Fender | This is getting worse |
| 15:32.01 | ganapathi | telecaller -> SIP User -> yate -> SIP trunk -> asterisk - > ISDN PRI -> Customer. |
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| 15:32.40 | [TK]D-Fender | caller starts the call. Should asterisk answer THAT call and force ringing to them when calling out to Yate as the next step? |
| 15:32.59 | ganapathi | yes. |
| 15:33.20 | [TK]D-Fender | Then go do it there |
| 15:33.26 | ganapathi | Yate dialing original customer call through ISDN PRI |
| 15:33.47 | ganapathi | ok. |
| 15:34.13 | [TK]D-Fender | At this point * will alsways consider the Telecaller leg to be answered in CDR and the phone will consdier it that way as well (answered/unanswered call lists affected, etc). |
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| 15:35.41 | ganapathi | Consider the position in reverse manner by this flow. telecaller -> SIP User -> yate -> SIP trunk -> asterisk - > ISDN PRI -> Customer. here telecaller register to yate and dialing a call then it's ringing to asterisk trunk then asterisk dial out from asterisk side. |
| 15:36.05 | ganapathi | here in dial plan if i answer then would work rgt ? |
| 15:37.20 | [TK]D-Fender | The closest side that answers between your Telecaller & PRI should be responsible for the tone. |
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| 15:38.16 | ganapathi | yes. i understand now by your response. |
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| 16:42.41 | igcewieling | Some days I wonder if I should move to NYC. |
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| 17:57.39 | mahafyi | Hello, could you recommend a reliable ISDN PRI service provider in California. We would cross connect to a colocation in a data center. |
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| 18:00.31 | igcewieling | That doesn't make much sense unless your COLO is in California |
| 18:01.36 | mahafyi | The colo is in california, probably we will go to coresite |
| 18:02.44 | igcewieling | I suspect your best bet for a TDM PRI would be the local ILEC. Be careful nobody sells you a "SIP PRI" |
| 18:04.20 | mahafyi | the data centers aren't putting us on the the providers. hence, i'd like to ask here for any reference. and right, we do not wand a pri thats connecting via bandwidth.com or something to pstn, lol. thanks for that advice. |
| 18:19.28 | mahafyi | ok sorry to repeat, where can find a directory of the carriers who can provide the isdn pri in los angeles, california? |
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| 20:11.15 | wr | anyone knows if datagram rdis ldap can transport frame relay? |
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| 21:03.15 | sicelo | are there any expected adverse effects from setting `strictrtp=no` in rtp.conf? seems to be the only reliable way for me so far to get my SIP client working well with asterisk |
| 21:05.32 | file | it means that no RTP source checking is done, so if someone sent RTP traffic to a port that was in use then they could inject media or get media sent to them (if symmetric RTP is in use) |
| 21:07.46 | sicelo | i've got this sip client which, even though it's bind NAT, on the same LAN as asterisk, seems to talk to asterisk using the WAN IP |
| 21:09.00 | sicelo | worked on it yesterday, and [TK]D-Fender helped me with some of the config, which improved it. but the issue was still there - intermittent, i may say. but with this setting, it really seems to work well all the time |
| 21:09.05 | sicelo | *shrug* |
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| 21:10.13 | sicelo | here's my sip.conf (the client uses 6001) - http://paste.debian.net/1044000/ |
| 21:14.42 | sicelo | this is me making an echo test, http://paste.debian.net/1044002/ |
| 21:18.26 | sicelo | so i hear the sound late,(lines 194/195). |
| 21:19.10 | sicelo | any better way to solve it (besides strictrtp=no)? maybe something to fix in my sip.conf .. not sure |
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| 23:34.02 | jameswf | Wow somehow they managed to make voip-info even worse than before |
| 23:34.32 | file | jameswf: inconceivable |
| 23:35.12 | jameswf | take all the stale information in the world.... convert it to a static wordpress site then wrap it in a 3cx advertisement |
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