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almoondsllc

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About almoondsllc

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  1. Is there a way to change the chipers Vodia offers? I'm trying with pipedream, things are going better... [9] 20:08:47.488 Initialize TLS connectionⓘ [9] 20:08:47.489 Last message repeated 2 timesⓘ [9] 20:08:47.489 HTTP 18.235.89.39: Send Client Hello(03035F97..00020017)ⓘ [9] 20:08:47.574 HTTP 18.235.89.39: Receive Server Hello(03035153..03000102)ⓘ [9] 20:08:47.575 HTTP 18.235.89.39: Receive Certificate(0012D700..5DF4038C)ⓘ [9] 20:08:47.575 HTTP 18.235.89.39: R
  2. Handshake Failure... [9] 19:57:25.204 Initialize TLS connectionⓘ [9] 19:57:25.204 Last message repeated 2 timesⓘ [9] 19:57:25.204 HTTP 149.154.167.220: Send Client Hello(03035F97..00020017)ⓘ [5] 19:57:25.209 HTTP 149.154.167.220: Alert Fatal (2): Handshake failure (40)ⓘ [7] 19:57:25.209 https:api.telegram.org:443: TCP disconnectⓘ [7] 19:57:25.209 https:api.telegram.org:443: Return code 500ⓘ
  3. Hello, I'm trying to receive Telegram message on my bot when a new call comes into the system I'm using HTTPS POST JSON according to telegram https://core.telegram.org/bots/api#making-requests So in Vodia -> Advanced ->Action URL I put: ** When a new call comes in ** Authentication metod: None Method: POST URL: https://api.telegram.org/bot<mytoken>/sendMessage Encoding for the message body: JSON Message Body: {"chat_id": "mychatid", "text": "This is a test from curl", "disable_notification": true} But it doesn't work, some h
  4. I think Vodia should say clearly if Zoho CRM/Phonebridge is supported or not. I got the enterprise version for the Zoho CRM integration, according the link https://doc.vodia.com/zoho_crm it should work but in fact it doesn't. I spent hours trying to get all this working and then Zoho support (I'm a Zoho one customer) told me this: Sorry to disappoint you. Please note that as of now you can only use the supported providers of Zoho. Please log-on to crm.zoho.com--> Click Setup--> Select Telephony under Channels and check the list if supported providers--> you can integrate any
  5. In fact I disabled the TLS on the phone and SRTP, the problem is that I still have 20 snom phones at this customer that now they are working with TLS, but I don't know for how long more it will be ok. How I can check the reason of the SRTP key negotation fails? If the solution is upgrading to a recent version of PBX, is the problem related to some Certificate expiration? If yes, how check when is the expiration? Just to be sure that nothing will happen until that date, thanks.
  6. Today I face out a very strange problem with a Vodia PBX (5.4.1 (Vodia mini PBX (Debian))) and a Snom M300/M65 set. If I enable TLS on the phone I had a very loud background noise before and during the call. All disappear if I use UDP as protocol on the phone instead of the UDP. How that is possible? I attached the audio file I recorded. WhatsApp Ptt 2020-07-28 at 09.46.02.ogg
  7. I started to play with the option in Hunt Group, From-Header and I realized that also with the just "Calling party name (CMC)" I got what I need: First line: <calling number> or <calling name - called number> if present in addressbook Second line: <called number> or <called name - called number> if present in addressbook This mean the call is delivered to the phone in different way if just like a simple extension or like hunt group. Anyway I solved my problem and I really thank you for your support. Any problem or need or request discussed in this forum
  8. It works! The interesting thing is this: Before the change, on the display of the snom I got this: First line of LCD: <caller id number> Second line of LCD: blank After the change you suggested: First line of LCD: icon of the phone ringing <caller id number> (<called number>) <<<---- according the new setup Second Line of LCD: icon of contact <name of the called number> <number of the called number> <<---snom automatically added So it's like then snom add a second line displaying the called nu
  9. The 50 DIDs on the system are working in this way: I have a sip trunk (ip auth) configured with the provider who assigned to me a GNR like +3905861234xx meaning all inbound calls from +390586123400 to +3905861234099 are routed in my pbx, so the pbx see coming call for +390586123400->99, at this point I should be able to intercept the DNID and display on the extension (snom d712) what's the +3905861234xx called (dnid) and who is calling (caller-id). I can of course assign the trunk sip receiving the 100 dnid to only one Hunt Group and then in the Hunt Group rules add a custom-heade
  10. About the hunt group and ACD, I saw the option you're talking about, the problem is that I've 50 DID and I can't create 50 hunt/acd group, I just need to modify how the caller id is displayed on the phone, if there is a way to "send just about anything in the SIP headers" I wish it would be also a way to do the same with inbound calls.
  11. My problems is showing on the Snom not only the callerid of the person calling but also the DID called (or the trunk name). So I don't know on which part of the setting of the trunk make changes: Number/Call Identification? Routing/Redirection? Shortly: caller +39370123456 calls the did +39058612345. I want to see on the snom (Snom 710) this: "+39370123456 (+39058612345)" or better, if I've mapped +39058612345. to "Google" in the addressbook: "+39370123456 (Google)"
  12. Yeah, I also noticed that, I solved pointing to 65.0.7 version that seems working even if I don't know if is a stable or beta or other version, I didn't see documentation about this release.
  13. Hello, I saw you added the "custom" option in SMS settings. Where to put Javascript code? Can you provider an example? Thank you. Regards, Alessandro Marzini
  14. Hello, I just installed in the internal lan on linux the last version, created some extensions and used on some pc on the lan. When I try to call or even setup audio, nothing happen. Is something related to https? (I'm using just http). Thanks. Alessandro
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