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| 15:55.14 | macaruchi | <PROTECTED> |
| 15:55.14 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.14 | macaruchi | [17:51] <macaruchi> When I use make menuselect I cant select CDR-PGSQL I get XXX |
| 15:55.18 | macaruchi | <PROTECTED> |
| 15:55.18 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.18 | macaruchi | [17:51] <macaruchi> When I use make menuselect I cant select CDR-PGSQL I get XXX |
| 15:55.38 | macaruchi | <macaruchi> I need to use Postgres for using with asterisk CDR |
| 15:55.38 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.38 | macaruchi | [17:51] <macaruchi> When I use make menuselect I cant select CDR-PGSQL I get XXX |
| 15:55.38 | macaruchi | <macaruchi> I need to use Postgres for using with asterisk CDR |
| 15:55.38 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.38 | macaruchi | [17:51] <macaruchi> When I use make menuselect I cant select CDR-PGSQL I get XXX |
| 15:55.38 | macaruchi | <macaruchi> I need to use Postgres for using with asterisk CDR |
| 15:55.39 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.39 | macaruchi | [17:51] <macaruchi> When I use make menuselect I cant select CDR-PGSQL I get XXX |
| 15:55.40 | macaruchi | <macaruchi> I need to use Postgres for using with asterisk CDR |
| 15:55.40 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.41 | macaruchi | [17:51] <macaruchi> When I use make menuselect I cant select CDR-PGSQL I get XXX |
| 15:55.41 | macaruchi | <macaruchi> I need to use Postgres for using with asterisk CDR |
| 15:55.42 | macaruchi | [17:51] <macaruchi> My question is i need to install ODBC for this or I can use directly libpq |
| 15:55.58 | macaruchi | Ooops!! Sorry |
| 15:56.02 | macaruchi | My bad |
| 15:57.13 | Samot | Yeah, totally unnecessary |
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| 16:40.48 | sahmed | Hello |
| 16:40.48 | sahmed | One question, Could we use stun binding request for rtp keep alive without ice/ice-lite session in asterisk? |
| 16:45.17 | [TK]D-Fender | macaruchi, Menuselect will tell you what dependency it is missing. |
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| 17:12.18 | macaruchi | PostgreSQL CDR Backend |
| 17:12.19 | macaruchi | <PROTECTED> |
| 17:12.43 | macaruchi | that is the dependecy pgsql(E) ? |
| 17:13.04 | file | the dependency is pgsql, which is an external library |
| 17:17.13 | macaruchi | I dont know that library let me find out |
| 17:18.09 | sahmed | Hello, Actually I want to know, is that practical to use stun binding for rtp keep-alive without ice/ice-lite session? |
| 17:19.22 | file | I have no idea, you'd have better luck consulting RFCs and posts to see what other SIP/RTP Implementors have come across with such a scenario |
| 17:20.48 | macaruchi | Everything about connect POstgres is ODBC driver |
| 17:21.49 | file | that is the core supported mechanism, but there is also the native module that you can use if you want |
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| 17:47.50 | Kobaz | chan_sip question... say you have peer foobar.com registered and you're sending invites... so you'll have INVITE 1234@foobar.com |
| 17:48.26 | Kobaz | is there a way to override that and use the ip address of the host instead.. so if foobar.com is 1.1.1.1, send INVITE 1234@1.1.1.1 |
| 17:48.54 | Kobaz | wihout changing the peer host... still want to register the peer by host name |
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| 18:06.37 | sibiria | Kobaz: there is the "externaddr" setting, if i understood your question correctly |
| 18:07.00 | Kobaz | nope that's not it |
| 18:07.09 | Kobaz | it's peer-specific |
| 18:07.16 | sibiria | you want to rewrite an incoming invite? |
| 18:07.19 | Kobaz | no |
| 18:07.22 | Kobaz | outgoing |
| 18:07.45 | Kobaz | say i have a DNS record sip-carrier.lan |
| 18:07.56 | Kobaz | my INVITEs are going out to 1234@sip-carrier.lan |
| 18:08.07 | Kobaz | sip-carrier does not like that.. and wants to see it's own ip in the INVITE |
| 18:14.29 | sibiria | ah, no, i don't think chan_sip can |
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| 18:15.20 | sibiria | but you can enforce the invitation to a specific host while still using a different peer iirc |
| 18:15.33 | sibiria | using the SIP/something//destination format |
| 18:16.12 | sibiria | it will still be partly static, however |
| 18:18.00 | Samot | You're forgetting something here... |
| 18:18.24 | sibiria | maybe you could use the "touser" trick in the dial string |
| 18:18.32 | Samot | You're focusing on the fact that it's an IP vs a FQDN. Either they those are still the _domain_ for the SIP URI. |
| 18:18.47 | Samot | Presenting a invalid domain will trigger protections, or should at least. |
| 18:19.24 | Samot | It doesn't matter that sip-carrier.lan resolves to the IP |
| 18:19.27 | sibiria | right, that's where touser@todomain in the dial string comes in |
| 18:19.34 | Samot | It's still the domain in the URI and will be invalid. |
| 18:20.53 | Slade | hmm. currently having a phone for a company being answered (in and out) just by a cellphone.. i wonder what a minor step up is, especially in VQ, and one that might allow multiple people to make/receive calls |
| 18:20.55 | sibiria | if i recall that format correctly, it's ! and !! |
| 18:21.13 | sibiria | SIP/something@blah!touser@todomain |
| 18:21.27 | sibiria | and !! would be for the from header unless my memory is off |
| 18:24.25 | Samot | Is there a reason you're using a custom FQDN in the host instead of what they provided? |
| 18:25.01 | Slade | ringcentral or something |
| 18:30.12 | Samot | Slade: You're describing a PBX so far. |
| 18:30.18 | Samot | So what is your actual question? |
| 18:31.18 | Slade | hmm, hard to phrase.. I'm just wondering what the nice incremental is, that isnt too labor intensive or expensive to make happen i guess. |
| 18:31.32 | Samot | Incremental what? |
| 18:31.41 | Samot | The calls are being forwarded to a cell phone... |
| 18:31.59 | Samot | Where do you want these calls to go? To some place that can allow multiple people to get that single call? |
| 18:32.15 | Samot | And call out through the same system instead of their cell phones? |
| 18:32.24 | Samot | Either via a deskphone or softophone? |
| 18:32.57 | Slade | everyone has the cellphone number, and they just call the cell.. and the phone is being passed around. its working, but sometimes theres a few people calling in, or sometimes someone wants to call out (as that number) while its being used. |
| 18:33.16 | Samot | So then you need a PBX system. |
| 18:33.22 | Samot | That is exactly what they are for. |
| 18:33.56 | Slade | indeed, either something remote virtual hosted, or get a physical one with a real land line.. or |
| 18:34.10 | Slade | there are lots of options at this juncture :) |
| 18:34.22 | Samot | There are quite a few options. We've discussed this with you before. |
| 18:34.49 | Slade | really? i'll check my logs |
| 18:35.11 | Samot | You can get a SIP trunk and setup a PBX |
| 18:35.27 | Samot | Whether that is in the cloud, or local on a VM or hardware. That's your call. |
| 18:35.35 | Samot | You can get a service like Ring Central.. |
| 18:35.44 | Slade | i remember discussing some specific solutions, just not sure if the conversation was ever about the cleanest way to make it happen :) |
| 18:35.49 | Samot | Pay per seat and just have phones and they are the PBX. |
| 18:36.03 | Samot | The cleanest way varies. |
| 18:37.57 | Slade | recommended PBX? |
| 18:38.06 | Samot | Uhm. |
| 18:38.16 | Samot | checks the channel |
| 18:38.20 | Samot | Oh yeah. |
| 18:38.50 | Slade | hmm. asterisk is a part of a pbx isnt it? not the entire solution/ |
| 18:38.56 | Samot | For you though, the GUI version will probably be better. So you can look at FreePBX or SwitchVox. |
| 18:39.23 | Slade | oh i thought both used asterisk |
| 18:39.39 | Samot | FreePBX does. |
| 18:39.56 | Samot | But those are GUI based. |
| 18:40.07 | Samot | Vs. Asterisk which is 100% config based with no GUI. |
| 18:40.47 | Slade | yes. messed with asterisk before.. everyone seemed to suggest not using it raw anymore because of all the extra features.. doesnt sound like thats the case according to you tho :) |
| 18:41.00 | Samot | I said _for you_ |
| 18:41.29 | Slade | oh, i meant in general. over watching last few years |
| 18:42.37 | Samot | There is a large part of the Asterisk community that would say raw Asterisk is the way to go. |
| 18:42.48 | Samot | Because it gives them the most control. |
| 18:43.12 | Samot | FreePBX is customizable but it does all the work for you. |
| 18:43.25 | Samot | So for someone who doesn't know the ins and outs of Asterisk... |
| 18:44.19 | Slade | https://www.sangoma.com/asterisk-vs-switchvox/ ah yea switchvox 'built on asterisk' slightly different wording |
| 18:45.19 | Slade | now to decide if i want virtual datacenter, or physical machine local |
| 18:46.17 | Slade | just a cheap vultr i guess :) |
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| 18:46.51 | file | Switchvox using Asterisk is an implementation detail, and one that all of its users don't care about or even know at times |
| 18:47.53 | Samot | Switchvox is also closed. |
| 18:48.17 | Samot | And costs $$$$ |
| 18:50.54 | Slade | i dont mind spending some money. tho they're probably going to do weird seat licenses or something |
| 18:55.15 | file | Switchvox is for if you don't want customization and just want something that's easy and just works generally |
| 18:55.57 | Samot | 1:34:50 PM <Slade> really? i'll check my logs < Holy hell I did and wow. |
| 18:56.51 | Samot | [2018-08-02 22:52:01] <Samot> Slade: FreePBX doesn't perform well in a container... |
| 18:57.04 | Slade | indeed |
| 18:57.18 | Samot | So, we've discussed this on and off for over a year. |
| 18:57.26 | Samot | SMS, cell phones, should it be a VM, hardware... |
| 18:57.31 | Samot | Which one to use... |
| 18:58.06 | Slade | still doesnt seem like a good softphone solution for on cellphones. they all seem like they're dying out |
| 18:59.15 | Samot | [2018-08-02 21:48:48] <Slade> Samot, cool. i'll look for a softphone that does that |
| 18:59.17 | Samot | [2018-08-02 21:48:55] <Samot> Bria. |
| 18:59.26 | Slade | yea i dont remember why i disliked bria |
| 19:02.12 | macaruchi | @file I dont fnd anything about that module native for postgres |
| 19:02.37 | macaruchi | Slade : Zoiper |
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| 19:12.54 | Slade | Samot, the thing that always holds me up switching to a PBX. losing SMS. but i just found flowsms... |
| 19:15.41 | igcewieling | Slade: All softphones suck. |
| 19:15.54 | Slade | igcewieling, my general conclusion as well |
| 19:16.00 | Slade | flowsms appears to be free too |
| 19:16.19 | igcewieling | Trenton's First Law of VoIP: All softphones suck. |
| 19:16.33 | Slade | voip.ms doesnt seem to have a similar service |
| 19:29.26 | aoeui | Bria isn't so bad |
| 19:30.09 | file | macaruchi: I'm not sure what you mean? the module itself is cdr_pgsql, but that is dependent on the Postgresql client library which is generally available in a package in the Linux distribution |
| 19:31.33 | igcewieling | Trenton's Second Law of VoIP: Some softphones suck less. |
| 19:33.34 | file | mobile softphones are hard to do well |
| 19:37.52 | file | (I'm like a broken record on that subject but I keep tabs on the Sangoma team for mobile and help them out, so I'm well aware of the madness in that area) |
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| 20:09.55 | drmessano | Bria's Push service is what makes it stand out. The call delivery is great |
| 20:12.30 | Slade | push service? |
| 20:16.34 | drmessano | Yep. Your phone registers to their proxy, which in turn registers to your PBX. When a call comes in, the proxy sends a push notification that "wakes up" Bria |
| 20:17.14 | drmessano | Not only does it allow Bria to "deep sleep" on the device, which extends battery life, it all but guarantees delivery |
| 20:17.59 | drmessano | Trying to maintain a SIP connection from a mobile device is about as dumb as trying to maintain an XMPP or IRC connection |
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| 20:21.18 | drmessano | It's disappointing to me that regardless of how many times or how often Samot and I recommend Bria, people still make the same arguments about battery life and maintaining a connection, when Counterpath has a REALLY great solution to it. |
| 20:21.32 | drmessano | I feel like I could cut and paste the last few lines from the last time this came up in here |
| 20:21.44 | Slade | interesting, googles claiming asterisk isnt easily multitenant, but freeswitch is |
| 20:23.18 | file | depends on what exactly multitenant means, Asterisk can certainly be configured to act in a multitenant fashion |
| 20:23.44 | Samot | I use Asterisk in a multi-tenant deployment |
| 20:24.00 | drmessano | That's another one of those equine necrophilia topics... |
| 20:24.12 | Samot | Again, this is also a topic that has been discussed. |
| 20:24.34 | Samot | VMs, SMS, Softphones, PBX vs UCaaS |
| 20:24.39 | Samot | For over a year. |
| 20:24.53 | Samot | Nothing has been done and we keep covering the same ground. |
| 20:25.06 | drmessano | Asterisk and FreeSWITCH are just as capable of being "Multitennant", but those looking for a GUI tend to lean towards FreeSWITCH, where a lot of that work has been done. |
| 20:25.13 | Samot | Slade: What has stopped you from making any sort of decision on this for almost 18 months. |
| 20:25.44 | drmessano | I think Thirdlane is the only Multitennant GUI I know of that covered Asterisk, and I think they threw their weight into FreeSWITCH |
| 20:26.11 | Slade | <PROTECTED> |
| 20:26.17 | drmessano | But there literally is no Asterisk Vs FreeSWITCH multi-tennant argument |
| 20:27.00 | Slade | drmessano, https://www.whichvoip.com/articles/freeswitch-vs-asterisk.htm last one i read was in here |
| 20:27.50 | Slade | Samot, slow moves and conflicting priorities. doing stuff that makes sense slowly. have some stuff moved onto flowroute, etc. |
| 20:28.26 | drmessano | That's a table on a site from some obscure provider that probably doesn't understand Asterisk or FreeSWITCH |
| 20:28.29 | drmessano | Source matters |
| 20:29.24 | drmessano | They're literally not even making an argument one way or the other, just representing it with a checkbox in a chart |
| 20:29.37 | drmessano | Find some actual discussion on the topic |
| 20:31.31 | drmessano | <PROTECTED> |
| 20:31.32 | drmessano | Sucks N Y |
| 20:31.36 | drmessano | ^ chart |
| 20:36.11 | Slade | i wasn't arguing for its accuracy. its just what the googles say |
| 20:37.16 | drmessano | Nor was I... I was arguing that what you selected wasn't even an actual argument. It was a checkbox on a site |
| 20:37.17 | Samot | It's also irrelevant. |
| 20:37.48 | Samot | Multi-Tenancy has no bearing on what you are doing. So it's a non-factor. |
| 20:39.16 | drmessano | Personally, I just don't deal in rumors |
| 20:39.21 | drmessano | Like "I heard Asterisk is slow" |
| 20:39.42 | Samot | Pfft. |
| 20:39.42 | drmessano | "Slow in what way?" "Google says it's slow. FreeSWITCH is faster" |
| 20:40.02 | Samot | You deal in rumors. Just only with the women you decide to date. |
| 20:40.13 | Samot | If there aren't any good rumors, forget about it. |
| 20:40.22 | drmessano | That's not even an argument or a discussion, it's repeating incomplete information that any random asshole could post on a blog or website |
| 20:41.08 | drmessano | "Asterisk can't multitennant. I see a lot on Google about it". OK, lets have a discussion. Cite a source, lets see their argument. |
| 20:41.23 | drmessano | Checkboxes. JFC |
| 20:41.52 | drmessano | VoIP is slow [ X ] Yes [ ] No |
| 20:42.01 | drmessano | OMG Proof |
| 20:42.08 | Samot | Well funny thing is... |
| 20:42.27 | TandyUK | scopserv is a full fledged multi tenant gui for asterisk too fwiw |
| 20:42.27 | Samot | I'm already dealing with an actual project that has gone on for 18 months because no one will make decisions. |
| 20:42.35 | Samot | And they want to talk about the same thing in circles. |
| 20:43.06 | drmessano | TandyUK: Impossible. I saw a red mark on a website that said Asterisk can't multitennant |
| 20:43.14 | TandyUK | rofl |
| 20:43.23 | TandyUK | and ofc its slow |
| 20:44.18 | TandyUK | the way i describe the difference betwee asteris kand freeswitch (which may not even be relevant any more) is... |
| 20:45.08 | TandyUK | Asterisk is the grandfather of VOIP, and was designed (and carries forward a lot of thinking) from when analog was the only way to do voip, consquently its very much 'one server' based, and failover tends to be an active/standby type |
| 20:46.01 | Samot | OK, wait. |
| 20:46.02 | TandyUK | Freeswitch otoh, is much newer and treats a call like a stream of packets, but can more easily offload how each packet is processed, so you can achieve active/active failover/fault tolerance a lot easier |
| 20:46.10 | Samot | Since when was analog the only way to do voip? |
| 20:46.16 | TandyUK | bear in mind ive been using that for maybe 10 years now |
| 20:46.17 | Samot | Since VoIP means Voice over IP |
| 20:46.27 | TandyUK | sorry , when analog was the only way to make calls |
| 20:46.39 | Samot | But they weren't. |
| 20:46.59 | Samot | The introduction of SIP and other VoIP techs made that possible. |
| 20:48.08 | TandyUK | theres strong arguments for using both depending on what the application is tbf, with kamailio or similar sitting in front |
| 20:48.46 | Samot | Now you're limited it to OSS. |
| 20:48.54 | Samot | limiting* |
| 20:54.25 | TandyUK | how am i limiting it? |
| 21:01.57 | Samot | Because not everyone uses FOSS |
| 21:03.13 | Samot | And there are certain non-FOSS things that have things that FOSS stuff doesn't. |
| 21:05.44 | *** join/#asterisk BakaKuna (~user@ip56509c03.direct-adsl.nl) |
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| 22:03.44 | igcewieling | $4,395 for a VMware image of Adtran n-Command hardware manager. |
| 22:07.03 | electronic_eel | just once or per year? |
| 22:14.12 | Kobaz | Samot: they provide an ip, straight up ip |
| 22:14.20 | Kobaz | Samot: so in case it ever changes, it's nice to use a hostname |
| 22:14.44 | Kobaz | SIP/something@blah!touser@todomain ... i'll try that |
| 22:22.44 | Samot | Kobaz: But my point still stands. |
| 22:22.58 | Samot | In SIP that would be the domain portion of the SIP URI |
| 22:23.16 | Samot | So just slapping any domain in there makes it invalid. |
| 22:23.41 | Samot | Like you could set the host to the IP and then set the outbound proxy to the domain |
| 22:24.24 | Samot | It would still format the RURI with the IP but send the request using the outbound proxy setting. |
| 22:24.29 | Kobaz | ah |
| 22:24.31 | Kobaz | okay |
| 22:24.53 | Kobaz | so i'm in the right place then, i started digging into chan_sip in transmit_invite() |
| 22:25.06 | Kobaz | i'm going to add an option for the sip domain to use for invites |
| 22:25.28 | Samot | I'm not sure why you are wasting time on this with Chan_SIP. |
| 22:25.38 | Samot | It's deprecated. |
| 22:33.09 | igcewieling | PJSIP seems more complicated, the lack of the stupid user/peer/friend model more than makes up for any added complexity in the config file. |
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