Scott1234

Yep just poking around.

Just wondering the story with what happened with this, seems to be gone? Vodia Billing with Stripe

Sharing is caring. Just create a dial plan for each state replacing +613 with +612 etc. And or remove + depending on what you are doing. 2;o-trk1;;000;+61000;;false 20;o-trk1;;^([0-9]{8})@;+613*;;false 22;o-trk1;;^0([0-9]{9})@|^61([0-9]{9})@;+61*;;false 24;o-trk1;;^([18,13][0-9]{9}|13[0-9]{4})@;+*;;false 80;o-trk1;;+*;+*;;false 82;o-trk1;;0011*;+*;;false

Inter domain dialling multi-tenant. As I was not able to find good doco when attempting this, I have made my own notes to share. Obviously special considerations are needed for overlapping extension domains, which the standard doco talks about. Ideally, I would like a way to only allow the alias/DID to be matched on the loop back not extension, mainly so when I dial the on-net tenant DID it goes internally, I don’t necessarily care about extension number to extension number but an option to pick would be nice. But I might be able to do that with, Use a list of expressions option for the inbound side matching. Note, Using global tenant dial plans This does not leverage a "try loop back" dial plan at all, not sure the danger but seems to be fine. No need to disable Loopback detection What I did, Create localhost domain Create trunk in localhost domain called on-net, sip gateway no registration, Outbound Proxy = sip:127.0.0.1:5060 (or pbx custom sip port) with the following settings. aadr: 127.0.0.1 analog: false ani_emergency: ani_regular: area_code: bcp: behind_nat: false cid_update: cobusy: 500 Line Unavailable code: codec_count: codec_lock: false codecs: codest: contact_hdr: country_code: dial_extension: dialplan: dir: dis: false dtmf: false dtmf_mode: earlymedia: true expires: 3600 failover: except_busy fraction: 128 from_source: ppi from_user: glob: global: true hcv: hd: hf: {from} hpai: hppi: hpr: hrpi: hru: {request-uri} ht: <{request-uri}> ice: false icid: identity: ignore_18x_sdp: true info_agent: false interoffice: false minimum: 10 name: on-net other_headers: outbound_proxy: sip:127.0.0.1:5060 pidflo: false prack: false prefix: redirect: false reg_account: reg_display: reg_keep: reg_registrar: reg_user: remote_party: request_timeout: require: reregister_dns: false rfcrtp: false ring180: false rtcpxr: false rtp_begin: rtp_end: sdpreq: send_email: sip_port: spam: false stir: t38_enabled: false teams: false tel: true trusted: true type: gateway use_epid: false use_history: false use_sdes: use_uuid: false user_defined_hdr: wrtc_dest_name: wrtc_dest_number: Edit your dial plans and place below emergency but above your normal region dial plans. 15;on-net;;*;*;;false As trunk on-net is set to fail-over except busy if no internal match is found it will flow as per the rest of the dial plan.

After further testing the only way I could get loop back to work when the base system was on non-standard SIP ports was to also add the original ports back in to the "SIP Settings" page. Even with specifying the 127.0.0.1:customport on the loopback trunk to use the nonstandard port it would not work. ¯\_(ツ)_/¯ Edit - Got it working on the standard port. I will post some doco as I was not able to find much to get this working.

Obviously due to the lack of response.. so when are you going to support Teams integration better?. Add on's, plugins, integrations all ways seem to be half baked in my experience with the pbx. I have a list of a lot more things wrong with Teams integration...

@Vodia PBX

I have only noticed one thing now that I have had this running out in the wild for a while. When using transfer.semi_attend_tran_enable = 0 Along with, transfer.dsskey_deal_type = 1 When switching it to a blind transfer during ring back as the call was started as a attended transfer it will trigger a missed call e-mail from the pbx when it transitions the call to blind transfer, which is annoying as you get a false positive of a missed call. Not sure if anything could be done @Vodia PBX? Maybe an adjustable missed call timer, so if less than say 2 seconds don't bother sending missed call.? The PBX seems to send a missed call right away even with a split-second direct call/hang up/cancel .

Did you forget to add the, to the pnp_yealink.xml ? I could not see on the web UI or in the xml, after upgrade. I see the voice country tone settings made it wooo

@Vodia PBX Another question to do with Customer Trunk headers. Are there any undocumented custom trunk headers that would allow us to make use of the "Other DNS address" setting of a domain, when constructing the header?

Also, to add to this. @Vodia PBX I noticed if a Call Q member is using the DND function of their phone the Q treats them as logged out as the agent login activity will reflect the use of DND. Which is great and expected. What's missing, Could the system put a check in place that's linked to the Number of logged in agents required to allow log out to Deny the use of DND? in that instance? A busy customer managed to have all of their agents on DND, thinking someone else was covering. I know what you're going to say , use the login log out function... But the login logout function is a bit more cumbersome than simply using DND especially if the agent is in more than one Q, I am all about streaming the user experience, the least steps possible. Also, BLF sync for single login logout BLF gets lost if there are internet glitches/outages the phone comes back the status of the login/out will be unavailable until next pressed which is annoying, might be a Yealink thing but who knows, I am not sure how it maintaining the BLF sync I do notice single login/out BLF buttons use some kind of special star code *60 that is not documented in the star codes section.

I am not sure what to say..... I will come back to this and post with more detail. when I have some spare time to try to explain what I am saying. As it does not work as intended with the default yealink_common settings, but I will make sure I document and capture everything I am saying. To try show you. It's not like I am making this up ....

I am not sure you are understanding what I am saying?? Here is an example you can test. Try and set this in Yealink General, with your default yealink config, key examples in bold. ##STUN NAT## account.1.nat.rport = 2 account.1.nat.nat_traversal = 1 static.sip.nat_stun.enable = 1 static.sip.nat_stun.server = stun.server static.sip.nat_stun.port = 3478 or ##daily autop for button renames## static.auto_provision.weekly.enable = 1 static.auto_provision.weekly.begin_time = 0:00 static.auto_provision.weekly.end_time = 3:00 static.auto_provision.weekly.dayofweek = 0123456 And you will see it won't apply them because the base file has them defined, below where the {parameter yealink-general} section is, hence the last occurrence wins, being. account.{lc}.nat.nat_traversal = 0 auto_provision.weekly.enable = 0

Yeah but I thought the idea here is to not have to edit the base file, which I am doing and I have removed {parameter yealink-general} from where it sits normally and put it below {post-process-yealink} due to what I have mentioned. I am saying if you simply move its location as part of the base pbx setup then we can be sure that any added changes to {post-process-yealink} will get applied and over write other defined settings. As yes it does seem to process it based on reoccurrence and over write that setting, even though the structure of the file generated shows two lots of commands with one being at the bottom, the last occurrence wins it seems. I guess to keep the config clean it would need a way of matching and merging/updating the same statements.

I can tell you thats not whats going on with it where it is. Hence why I have posted it. Only resolved with it at the bottom.

Yep, I program into general prams for that reason, but you guys need to move the general parameter include statement to the end of the Yealink provisioning template so that if we want to make a change to something that exists in the config below where it includes the parameters now it's not overwritten by the base again.

or IP whitelist

Here are my Yealink tweaks that might help you, note some only work when you modify the base yealink config and move the general pram include statement to the end of the file. I have found running with this setup I get way less call errors and failed alerts from the Yealink YMCS management portal. ##DNS CONTROL## (Never trust a customers DHCP DNS set your own..) static.network.static_dns_enable = 1 static.network.primary_dns = x.x.x.x static.network.secondary_dns = x.x.x.x ##STUN NAT## (less random errors noted in YMCS) account.1.nat.rport = 2 account.1.nat.nat_traversal = 1 static.sip.nat_stun.enable = 1 static.sip.nat_stun.server = stun.server static.sip.nat_stun.port = 3478 ##Remove SIP server 2 duplicate## account.1.sip_server.2.address = %NULL% ##DM YEALINK## (YMCS great for remote management) dm.enterprise_id = enterprise code dm.site_id = site code ##LOCAL TONES## (Make true tones for your country) voice.tone.country = countryname ##Dial Now Delay## (When using PNP proper dial plan) phone_setting.dialnow_delay = 2 ##LONG KEY LABLES## (More button realestate is welcomed by all) features.config_dsskey_length = 1 ##Fix DSS Trans with pbx call back## (Fixes transfer so can make use of pbx transfer reminder) transfer.semi_attend_tran_enable = 0 ##CustomRingTones## (I found I still had to upload my own to the phone) ringtone.url = my custom tones ringtone.url = my custom tones ##daily autop for button renames between midnight and 3am random## (Daily random sync so if customer puts in jobs for button name updates all phones will sync to download out of hours) static.auto_provision.weekly.enable = 1 static.auto_provision.weekly.begin_time = 0:00 static.auto_provision.weekly.end_time = 3:00 static.auto_provision.weekly.dayofweek = 0123456 ##Phone web timeout## (Sick of being logged out to soon when doing remote support) features.relog_offtime = 30 ##Fixs for pickup from blf and parks when using no outbound proxy ## features.call_park.park_mode = 1 features.pickup.direct_pickup_code = *87 features.blf_pickup_only_send_code = 0 account.1.dialoginfo_callpickup = 0 ##BLF PAGE TIPS## (When monitoring more BLF's and you get the virtual page key, the phone will flash the page that has the monitored blf blinking) phone_setting.page_tip = 1

You don't really need to set dir settings url unless you build a custom xml. yealink-favorite.xml in phone pnp covers directory settings url, you will figure out what its doing by looking at that. DSS Key Type, I all ways use "1" as this has the best user experience on Yealink. When using mode 1, pressing a monitored BLF key while in call will result in a Attended transfer, so you can be in the call then press DSS BLF key and the caller is put on hold automatically and a new call is started to the other extension then you can talk and hang up to transfer or press transfer soft key. Default operation is blind transfer, which means you have to manually hold the call then start a new call and press the BLF button, not ideal, too many steps. With mode one you can get the call to the other party in two moves vs three or four, I forget how many extra steps default mode made it but it feels awkward. If you combine DSS Key Type 1, with my other post post on Yealink and transfer reminder you will get the best operation. I also find better stable operation of using park mode in FAC mode on Yealink. I only do yealink's so I know a fair bit on them, AMA lol. It was once snom but prices here are way too high.

While reading some other Yealink doc's I came across doco on something that could work better as a hot desking function, all though not intended for that. As I too have had issues with 'hot desking' which for functionally reasons I need to do it not using the pbx's vison of hot desking. Yet to investigate fully, but my thinking is you could use PIN code AutoP where you have a base common config, then the pin CFG files (hot desk users) stored alongside in the backend of the pbx. Yealink SIP IP Phones Auto Provisioning Guide V1.0.pdf Refer to, Auto Provisioning via Activation Code & Auto Provisioning via PIN Code, then you can treat each user load like it was a separate device with its own config. But without yet testing not sure if the user workflow trying to do it this way would be acceptable, its on my list at some point to try out.

Think's I see the android was updated, I will check it out and comment back.

So, did you ever get to messing around with this? The Teams admin GUI now caters for inputting the DID / extension separately which makes it present neatly on the team's client. so instead of setting, the three-digit extension as the "DID" we should be setting that as the extension as per above, but the trunk would need a option for it.

When started from the PBX side so, external call. It will use SRTP and this is what the SDP looks like, and the Protocol will be marked as SRTP. And no SRTP fail alerts from Yealink DM/RPS system. PBX -> Phone Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 774385152 774385152 IN IP4 XXX.XXX.XXX.XXX Session Name (s): - Connection Information (c): IN IP4 XXX.XXX.XXX.XXX Time Description, active time (t): 0 0 Media Description, name and address (m): audio 51428 RTP/SAVP 8 101 Media Attribute (a): crypto:1 AES_CM_128_HMAC_SHA1_32 inline:oP4MFtsvAaIRFPjMEY+vuaRhV0IS+Wu95uRyaiSQ Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): ptime:20 Media Attribute (a): rtcp-xr:rcvr-rtt=all voip-metrics Media Attribute (a): sendrecv [Generated Call-ID: 42ee1295@pbx] Phone -> PBX Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 20009 20009 IN IP4 XXX.XXX.XXX.XXX Session Name (s): SDP data Connection Information (c): IN IP4 XXX.XXX.XXX.XXX Time Description, active time (t): 0 0 Media Description, name and address (m): audio 12440 RTP/SAVP 8 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): ptime:20 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ttCdLX6ZvWf7NoUibr+tyZ4d2pQx8dtMhXb98Tdn Media Attribute (a): sendrecv Media Attribute (a): rtcp:12441 [Generated Call-ID: 42ee1295@pbx] From Yealink Site, https://support.yealink.com/en/portal/knowledge/show?id=5acd46f5b10e6e087e86b614

The SDP looks like this with a call starting from the handset. But then proceeds to send as non SRTP. Phone -> PBX Frame 1: 1356 bytes on wire (10848 bits), 1356 bytes captured (10848 bits) Ethernet II, Src: 00:00:00_00:00:00 (00:00:00:00:00:00), Dst: 00:00:00_00:00:00 (00:00:00:00:00:00) Internet Protocol Version 4, Src: XXX.XXX.XXX.XXX, Dst: XXX.XXX.XXX.XXX User Datagram Protocol, Src Port: 12270, Dst Port: XXXX Session Initiation Protocol (INVITE) Request-Line: INVITE sip:XXX@XXXXX.COM:XXXX;user=phone SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 20008 20008 IN IP4 XXX.XXX.XXX.XXX Session Name (s): SDP data Connection Information (c): IN IP4 XXX.XXX.XXX.XXX Time Description, active time (t): 0 0 Media Description, name and address (m): audio 12438 RTP/AVP 8 0 107 9 101 102 Media Attribute (a): rtcp:12439 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): fmtp:107 sprop-maxcapturerate=48000; maxaveragebitrate=40000; maxplaybackrate=48000; useylrtx=1; useinbandfec=1 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): ptime:20 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtpmap:102 telephone-event/48000 Media Attribute (a): fmtp:102 0-15 Media Attribute (a): crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Qd1a3pYU0wuRVFUYIHgjag/i7U05sJy+eNQBAOkr Media Attribute (a): crypto:2 AES_CM_128_HMAC_SHA1_32 inline:rq9ThvgXz6vkwdbFZCIfCdR6qGcowDhvNJvTBPKv [Generated Call-ID: 0_911050508@192.168.1.33] PBX -> Phone Frame 3: 1020 bytes on wire (8160 bits), 1020 bytes captured (8160 bits) Ethernet II, Src: 00:00:00_00:00:00 (00:00:00:00:00:00), Dst: 00:00:00_00:00:00 (00:00:00:00:00:00) Internet Protocol Version 4, Src: XXX.XXX.XXX.XXX, Dst: XXX.XXX.XXX.XXX User Datagram Protocol, Src Port: XXXX, Dst Port: 12270 Session Initiation Protocol (200) Status-Line: SIP/2.0 200 Ok Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 323925493 323925493 IN IP4 XXX.XXX.XXX.XXX Session Name (s): - Connection Information (c): IN IP4 XXX.XXX.XXX.XXX Time Description, active time (t): 0 0 Media Description, name and address (m): audio 53990 RTP/AVP 8 101 Media Attribute (a): crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NoxP2O73jCrJ6flyRiTHneE9eC9Ex3wDjB0npLrR Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): ptime:20 Media Attribute (a): rtcp-xr:rcvr-rtt=all voip-metrics Media Attribute (a): sendrecv [Generated Call-ID: 0_911050508@192.168.1.33] Key's are being offered, but not being accepted. Any ideas on what to look at ? I really want end to end encryption. The phones all talk over TLS no problems, even if you say use only trusted certs on the phone that all works.

The iPhone app is great, but the android one seems to have fallen behind, Is this going to be brought in line with iPhone? as uniform app training between the OS's would be great. I see its last update was Dec 22.