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| 12:38.27 | emsjessec | is it possible to dial numbers on a VOIP handset by playing tones into the phone? |
| 12:38.55 | Samot | What do you mean by that? |
| 12:39.35 | emsjessec | DTMF tones |
| 12:41.32 | Samot | And how would you play those tones into the handset? |
| 12:42.18 | Someone_Else | whenever you dial out, asterisk chooses a outbound route; is there a possibility to hook in before that selection and select the outbound route manually (based on some result)? |
| 12:42.39 | emsjessec | Samot, a computer |
| 12:42.45 | emsjessec | holding the phone to the speaker |
| 12:43.16 | Samot | No. |
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| 12:53.10 | emsjessec | i have a different question |
| 12:53.29 | emsjessec | if a PBX has a FXS1 port, can an analog phone be connected and dial out? |
| 12:54.27 | Samot | Yes. |
| 12:54.53 | file | Someone_Else: there is nothing built in, it's all configured in dialplan and Asterisk does what it is told there |
| 12:54.53 | emsjessec | when I try it says "You are not allowed to dial this number." |
| 12:57.14 | Samot | emsjessec: OK, that doesn't really help. |
| 12:57.27 | emsjessec | i opened a ticket with GrandStream PBX |
| 12:57.39 | emsjessec | how does Asterisk differ from a GrandStream? |
| 12:58.13 | Samot | Wait, this is a GS PBX? |
| 12:58.24 | emsjessec | yes |
| 12:58.30 | Samot | Then we've got nothing for you. |
| 12:58.37 | Samot | Grandstream is a close sourced PBX. |
| 12:59.04 | emsjessec | i'm not asking about the GrandStream in particular |
| 12:59.18 | Samot | Well if this is where you FXS card is and giving you that error... |
| 12:59.21 | Samot | It's a GS PBX issue. |
| 12:59.29 | emsjessec | if VOIP handsets are working, would it be likely that a VOIP to analog adapter would work? |
| 12:59.39 | Samot | Sure. |
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| 13:00.54 | emsjessec | can you recommend a VOIP service for a home phone system? |
| 13:02.31 | emsjessec | does an analog signal require a lot more bandwidth than a VOIP signal? |
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| 15:51.22 | Helenah | Is there anything specific which I need to do to forward asterisk box which is behind a NAT? |
| 15:52.08 | mTeK | Forward 5060-5061 |
| 15:52.22 | mTeK | If you have external phones |
| 15:52.40 | Helenah | I did, however when I scan my public IP from outside the LAN, the ports aren't avail. |
| 15:52.41 | mTeK | turn off sip alg or sip helper on router |
| 15:52.50 | Helenah | I did that already. |
| 15:53.21 | [TK]D-Fender | Maybe you scan wasn't valid <- |
| 15:53.23 | mTeK | Then do didn't open the ports correctly or your provider is blocking them. |
| 15:53.30 | Helenah | I think I changed external_media_address and external_streaming_address. |
| 15:53.36 | mTeK | yes thats correct |
| 15:53.51 | Helenah | I'll get onto my ISP. |
| 15:53.59 | Helenah | Thank you very much! |
| 15:54.02 | mTeK | CHeck fireall in modem |
| 15:54.38 | mTeK | Make sure your using bridge mode or IP passthough and not DMZ to your router. |
| 15:54.59 | mTeK | DMZ mode can cause nat sip problems |
| 15:55.44 | Helenah | So I can't use port forwarding? |
| 15:56.03 | mTeK | If your using the modems firewall then yes |
| 15:56.20 | sibiria | you need to either way, if the asterisk box is on a private network |
| 15:56.24 | sibiria | this is what NAT is for |
| 15:56.39 | Helenah | My modem is a DM200 which is just a dumb modem, my Mikrotik router handles the PPPoE credentials. |
| 15:56.42 | mTeK | sibiria: I think he is talking about external phones |
| 15:56.53 | Helenah | and my Mikrotik is the NAT. |
| 15:56.53 | mTeK | Ok |
| 15:57.31 | Helenah | Are you saying that I would need a firewall on the DM200? |
| 15:57.35 | mTeK | Then make sure your nat rule is correct and above any block rules in the filter tab |
| 15:57.58 | Helenah | All my other NAT rules are working fine. |
| 15:58.08 | mTeK | You can also make a dumy rule with just logging on for the filter and then you can see if the packets are hitting the firewall |
| 15:58.21 | mTeK | Make sure you forwared UDP |
| 15:58.27 | Helenah | I did. |
| 15:58.30 | mTeK | 5060 is UDP |
| 15:58.46 | Helenah | I'm only using 5061/udp. |
| 15:58.47 | mTeK | 5061 should be TCP... Think so |
| 15:58.52 | Helenah | Wait... |
| 15:59.04 | Helenah | Seriously? |
| 15:59.12 | mTeK | I don't use freepbx anymore so that was off the top of my head |
| 15:59.12 | Helenah | Is there a source for that information? |
| 15:59.24 | mTeK | Other vendors use 5061 for TCP only |
| 15:59.53 | sibiria | 5061 is for TCP. don't use UDP there |
| 16:00.01 | sibiria | it breaks conformance |
| 16:00.08 | Corydon76 | 5061 is for TLS-secured SIP |
| 16:00.16 | Helenah | So we are both halfway right. |
| 16:00.32 | Helenah | Apparently both ports 5060 and 5061 use both TCP and UDP. |
| 16:00.51 | sibiria | ah, yes, sorry of course |
| 16:00.58 | Samot | Actually, it doesn't matter. |
| 16:01.00 | Corydon76 | 5060 is fine for TCP. There's no conflict with UDP; the ports are specific to the protocol. |
| 16:01.02 | sibiria | 5060 TCP is the fallback for too large UDP packets |
| 16:01.04 | Samot | Those are the standard defaults. |
| 16:01.05 | mTeK | 5060 will never use tcp unless you;ve configured it that wayu |
| 16:01.17 | Samot | That doesn't mean you can't use them for something else. |
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| 16:01.53 | Helenah | But, I am best forwarding both transports UDP and TCP on those ports, right? |
| 16:01.59 | mTeK | Yes |
| 16:02.05 | Samot | Are you using TCP? |
| 16:02.11 | [TK]D-Fender | only is you're using them both |
| 16:02.13 | [TK]D-Fender | if* |
| 16:02.16 | Corydon76 | You are best using 5060 only unless you have configured TLS |
| 16:02.16 | Samot | There is no TCP fall back. |
| 16:02.17 | [TK]D-Fender | You forward what you use |
| 16:02.25 | [TK]D-Fender | look at what you configured |
| 16:02.41 | sibiria | you should allow TCP, or you'll break RFC-somethingsomething which mandates TCP for payloads above a certain size |
| 16:02.50 | Samot | Why? |
| 16:03.07 | Samot | If they are not using TCP for SIP then there is no need to allow TCP for it. |
| 16:03.15 | mTeK | You will reach the mtu before you hit that... |
| 16:03.41 | Samot | Helenah: Are you having issues? |
| 16:03.44 | sibiria | i don't know if they having incoming traffic from some external vendor. just pointing out what the standard asks |
| 16:03.50 | sibiria | if they are having* |
| 16:04.48 | Samot | Helenah: I use Mikrotik 100% for all my voice stuff. So are you having issues? |
| 16:05.57 | sibiria | mTeK: iirc the limit is moving, or 1300 bytes regardless of the agreed MTU |
| 16:06.15 | sibiria | something along the lines of 128 bytes within the MTU, or 1300 bytes |
| 16:07.03 | [TK]D-Fender | heads home |
| 16:10.42 | Samot | Helenah: ?? Because it's very rare to have to put NAT rules in for SIP on Mikrotik's. Very few cases require it. |
| 16:12.18 | Helenah | So... apparently I'm using TCP. But there is no Audio, I checked the RTP NAT rule and no traffic goes through it. |
| 16:13.14 | Samot | OK, so why are you using TCP? |
| 16:14.09 | Helenah | Not sure, I wondered that myself, but I thought I'd give it a try. I'll set it to UDP. Can it be set to both? What would the advantages of that be? |
| 16:14.31 | Samot | OK well first that is just the signalling. Second, RTP is always UDP. |
| 16:14.41 | Helenah | Bare in mind, I don't use port 5060. |
| 16:15.13 | Samot | Show your PJSIP endpoint config. |
| 16:20.02 | Helenah | Is port 5060 needed for an Asterisk setup? I'd rather it be TLS-only unless absolutely necessary. |
| 16:20.26 | sibiria | it's not a requirement that you stick to 5060 for unencrypted and 5061 for sips/srtp |
| 16:20.46 | sibiria | those are just the standard ports |
| 16:21.12 | Corydon76 | SRTP shouldn't be on 5061 anyway |
| 16:21.17 | Samot | Well standard for SIPS is one port higher than the SIP port. |
| 16:21.41 | Samot | So if SIP is 5080 than the standard states TLS should be 5081 |
| 16:22.49 | Helenah | My friend is setting asterisk without support for port 5060, she is using only 5061 so that all connections are TLS. Does she need to use asterisk with 5060 and 5061 ? |
| 16:23.59 | Helenah | or do I press the big reject button and tell her to make our teas - hahaha byeeee |
| 16:25.12 | sibiria | remember that just using 5061 for sip alone doesn't imply encryption, just as sips itself doesn't imply srtp |
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| 17:08.21 | jrun | what's the use of SSRC? |
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| 17:28.03 | jrun | when a single endpoint within an rtp session wishes to send multiple media streams, does it open a port for each? can it send all stream over the same port? |
| 17:30.38 | file | there's a draft for using a single underlying transport (port) for multiple streams, it is called bundle |
| 17:30.42 | file | by default it is separate. |
| 17:33.45 | jrun | does asterisk accept same ssrc on different rpt session at the same time? or is it more like that each rtp session has it's own namespace? |
| 17:34.19 | file | each RTP session has its own. |
| 17:43.18 | jrun | what is the most generic way of identifying rtp sessions? (port, ssrc) combination or that won't work? |
| 17:43.32 | jrun | actually how does asterisk do it? |
| 17:45.31 | jrun | file: this? https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-36 |
| 17:45.45 | file | that is the draft for bundle, yes |
| 17:47.01 | file | as for identifying generally port for older stuff... some filter based on learnt SSRC, or in the case of bundle you do SSRC and there's other RTP extensions that can potentially identify based on signaling communicated data |
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| 23:31.24 | life_of_e | Why would there be a nearly hour long delay in voicemail indicators (usually the MWI on a phone) after a voicemail is recorded? |