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| 00:20.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); 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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| 13:52.03 | igcewieling | is res_rtp_asterisk.so only for pjsip? : loader.c:583 load_dlopen: Error loading module 'res_rtp_asterisk.so': /usr/lib64/asterisk/modules/res_rtp_asterisk.so: undefined symbol: ast_pjproject_caching_pool_destroy |
| 13:54.26 | file | res_rtp_asterisk does RTP for chan_sip, chan_pjsip, and other things - it uses pjproject for ICE/STUN/TURN |
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| 13:57.11 | igcewieling | Any idea why it won't load? I'm not using pjsip. |
| 13:57.30 | file | res_pjproject.so isn't loaded? |
| 13:57.47 | file | if it was still built then res_rtp_asterisk would have built with usage of it |
| 13:57.50 | igcewieling | it needs to be loaded even if pjsip isn't used? |
| 13:58.26 | file | if it was built, yes, because PJSIP is a component of PJPROJECT - and res_rtp_asterisk uses other parts of it |
| 14:00.00 | igcewieling | file: thanks for the info. It helps a lot. |
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| 15:54.24 | lwlvl | When I ring inbound, the number (german number) has a local format (i.e. 015783844423). See SIP-From-Header here: https://bpaste.net/show/df14474e9b6f . The problem is, that this number is directed to my voip-phone, which gets the number without the leading 0 for getting out of the PBX. I think this is caused by the SIP-From-Header. Does anyone have suggestions how to solve this? |
| 15:55.42 | [TK]D-Fender | Go add it to the callerID |
| 15:56.18 | [TK]D-Fender | Caller ID comes in wrong -> GO CHANGE IT. It's your dialplan. |
| 15:56.59 | lwlvl | [TK]D-Fender, Set("SIP/sipconnect.sipgate.de-000000db", "CALLERID(num)=+4915700000143") in new stack |
| 15:57.30 | lwlvl | [TK]D-Fender, that's what I did. did I also Set the name? |
| 15:57.51 | lwlvl | sry...do I also have to set the CALLERID(name)? |
| 15:58.20 | [TK]D-Fender | name != number |
| 15:59.18 | [TK]D-Fender | You also said you needed another 0 ... then you show a sample where you ar adding all sorts of other stuff. |
| 15:59.29 | lwlvl | [TK]D-Fender, thnx, that solved my issue. I also had to modify CALLERID(name) |
| 15:59.34 | [TK]D-Fender | Your proposed fix there doesn't sound like your original request at all |
| 16:00.18 | lwlvl | [TK]D-Fender, yes...because I thought is was the sip-from-header causing the problem |
| 16:00.58 | lwlvl | [TK]D-Fender, another 0 or the +49...I decided to use the +49 |
| 16:01.45 | lwlvl | [TK]D-Fender, because my asterisk reformats the numbers when calling out to fit the "49XXXXXXXX" format of my provider.... |
| 16:01.54 | [TK]D-Fender | From Header is just a common source of callerid which is what can get applied to a call out from your PBX |
| 16:01.58 | lwlvl | and +49 is okay for it ;) |
| 16:02.23 | lwlvl | ok |
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| 19:47.04 | Kharos16 | Hey! |
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| 19:48.24 | Kharos16 | can i have some pointers? i'm trying to setup a small voip network but the server is not physically here while the router with the E1 plugged is, how do i route the E1 to the server so i can get calls from my main office to a branch where my server is? |
| 19:49.54 | [TK]D-Fender | put a gateway there. |
| 19:50.13 | [TK]D-Fender | E1 has to turn into SOMETHING to be sent out of that location. |
| 19:50.23 | [TK]D-Fender | Which would typically be a gateway to SIP, etc |
| 20:42.43 | Kharos16 | thanks |
| 20:43.32 | Kharos16 | i was trying to figure out a way that wouldn't need a voice gateway as my boss said he wouldn't be buying one |
| 20:43.41 | Kharos16 | but it seems we have hit a wall |
| 20:44.06 | Kharos16 | if anyone knows what can i reference or look for apart from a voice gateway i'll be eternally grateful to you |
| 20:45.46 | Samot | A gateway is your only choice is the PBX is not there. |
| 20:46.05 | Samot | Otherwise you need to have a PBX on-premises and use a E1/PRI card in the machine. |
| 20:47.02 | Samot | The router with the E1 is a full CPE, like an Adtran, that does the Internet and the voice. |
| 20:47.09 | Kharos16 | Ok, can the router i have the E1 card on function as a voice gateway? |
| 20:47.17 | Samot | No. |
| 20:47.25 | Samot | Why is the E1 card in the router? |
| 20:47.30 | Samot | Did the ISP provide it? |
| 20:48.23 | Samot | Is there a PBX on-premises now? |
| 20:48.23 | Kharos16 | i'm picking up this setup from a guy before me that quit apparently boss told him no when he talked about buying a voice gateway and told him to use one of the E1 cards on the warehouse and put it on a cisco router to function as a voice gateway |
| 20:49.00 | Kharos16 | No all i have is a remote virtualized server |
| 20:49.18 | Kharos16 | thats ok for voip but is a must that we can make calls on the pstn |
| 20:49.20 | Samot | Is this a new E1? |
| 20:49.51 | Samot | VoIP can make calls to the PSTN. |
| 20:50.02 | Samot | As long as you have a route/path to the PSTN. |
| 20:50.13 | Samot | In this setup the E1 is that path. |
| 20:50.19 | Kharos16 | I'm not sure about what you mean when you say new, the line is newly leased if thats what you mean |
| 20:50.34 | Samot | I mean was this E1 recently installed? |
| 20:50.42 | Kharos16 | Yes |
| 20:50.43 | Samot | For the purpose of voice services... |
| 20:50.44 | Samot | OK |
| 20:50.49 | Samot | This was done wrong. |
| 20:51.05 | Samot | If you have a PBX in the cloud, you should be using SIP. |
| 20:51.17 | Samot | And going with an ITSP that does SIP trunks. |
| 20:51.26 | Kharos16 | as far as i can see in the configuration files on the router yes |
| 20:51.59 | Kharos16 | Yes sir |
| 20:52.10 | Samot | What type of Cisco router is this? |
| 20:52.23 | Samot | Because in order to make it a "voice gateway" it must be able to do that. |
| 20:53.26 | Kharos16 | give me a moment to get a show version on that router |
| 20:55.45 | Kharos16 | cisco 2951 router |
| 20:56.12 | Kharos16 | C2951 Software (C2951-UNIVERSALK9-M), Version 15.4(3)M2, RELEASE SOFTWARE (fc2) |
| 20:56.39 | jpsharp | woo, top of the line stuff there. |
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| 20:58.25 | Samot | Well the 2900's are maintained until about 2020 |
| 20:58.33 | Samot | But they are EOL and no support. |
| 20:59.10 | Kharos16 | I have a correction to make, it is this router Cisco IOS Software, 2800 Software (C2800NM-IPVOICEK9-M), Version 15.1(4)M8, RELEASE SOFTWARE (fc2) |
| 20:59.11 | Samot | Oh no that goes a little longer. |
| 21:00.14 | Samot | Well you just took a step backwards. |
| 21:00.27 | Samot | That's completely EOL and no longer maintained. |
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| 21:01.40 | Kharos16 | I'm sorry i must work with what i'm given :( |
| 21:01.56 | Samot | That problem is what you were given is a big turd. |
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| 21:03.37 | Kharos16 | so there's really nothing else i can do but try to talk my way out of this one? |
| 21:05.08 | Samot | The 2800's are Cisco Integrated systems. |
| 21:05.15 | Samot | They run the Cisco Call Manager... |
| 21:05.32 | Samot | It can do voice but the router is the PBX/system. |
| 21:05.46 | Samot | It will not do a SIP trunk to Asterisk as far as I can see. |
| 21:05.52 | Samot | And if it can, good look. |
| 21:06.03 | Samot | luck |
| 21:06.29 | Samot | Whoever thought the E1 + a cloud based PBX was a good idea was completely wrong. |
| 21:07.06 | Samot | If there were SIP/ITSP providers that could provide service for this, they were the better choice. |
| 21:13.28 | Kharos16 | Well still thank you for your time :) |
| 21:13.35 | Kharos16 | guess i can only talk it up with the boss |
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| 21:22.03 | Samot | You have three options |
| 21:22.21 | Samot | 1) Get a gateway that will take the E1 and convert it to SIP so you can trunk to the PBX |
| 21:22.47 | Samot | 2) Put a PBX on-premises, use a gateway or get an E1 card to put in the PBX |
| 21:23.00 | Samot | 3) Dump the E1 and just get a SIP trunk for the PBX. |
| 21:23.17 | Samot | Either way, none of those options are going to be cheap. |
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| 21:30.59 | jpsharp | voip is scary for people. |
| 21:35.20 | drmessano | The biggest drawback to VoIP is that is requires IP transport, which goes over a network |
| 21:35.25 | drmessano | If it wasn't for that, boom |
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