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We have changed the strategy for Caller-ID. Instead of using the INFO messages that were pretty much special for snom phones, we use the P-Asserted-Identity now. This seems to work with all phones that we could test, including snom phones at least with a newer firmware.

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Just installed 5.2.0 - the caller ID works as expected which is great, however the received caller list is still showing the internal callers name and ext number.

 

Should the Caller ID also show the internal party that transferred the call? At the moment it shows both the 0423 XXX XXX and +61 423 XXX XXX numbers.

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For transfers it is really a tricky topic what to show. For the web interface, I think it is reasonable to just show the caller and the called party. We have added a very detailed new record type for those who need to know the details, which can be pushed out by HTTP/JSON. It will require an external server to pick up the CDR; but they will contain really everything.

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Guys,

 

As I could see - you simply ignoring the fact the inter-domain calls is broken. In 5.2.0 it is not working also.

 

Domain / routing part identified correctly, but for some reason - "SIP/2.0 501 Temporarily Unavailable"

 

When it will be fixed??? Very, very disappointing decision to ignore customers. We are thinking to massively extend the amount of licenses (add more companies) but inter-domain calls is a "must"

 

 

Calling from 3000 to 7013:

 

[9] 12:10:09.147 SIP: Resolve 815: udp 127.0.0.1 5060
[5] 12:10:09.147 PACK: SIP Tx udp:127.0.0.1:5060:
ACK sip:7013@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-9b7c161e71e7da39fc5da3f55e38443c;rport
From: "Maxim Shaposhnikov" <sip:3000@pbx.highperf.pro>;tag=1985252171
To: <sip:7013@pbx.navica.highperf.pro;user=phone>;tag=cb47e67e08
Call-ID: a35e627e@pbx
CSeq: 21764 ACK
Max-Forwards: 70
Contact: <sip:3000@127.0.0.1:5060;transport=udp>
P-Asserted-Identity: "Maxim Shaposhnikov" <sip:3000@pbx.highperf.pro>
Content-Length: 0
[5] 12:10:09.148 SIP: INVITE Response 501 Temporarily Unavailable: Terminate a35e627e@pbx
[5] 12:10:09.148 PACK: SIP Rx udp:127.0.0.1:5060:
ACK sip:7013@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-9b7c161e71e7da39fc5da3f55e38443c;rport
From: "Maxim Shaposhnikov" <sip:3000@pbx.highperf.pro>;tag=1985252171
To: <sip:7013@pbx.navica.highperf.pro;user=phone>;tag=cb47e67e08
Call-ID: a35e627e@pbx
CSeq: 21764 ACK
Max-Forwards: 70
Contact: <sip:3000@127.0.0.1:5060;transport=udp>
P-Asserted-Identity: "Maxim Shaposhnikov" <sip:3000@pbx.highperf.pro>
Content-Length: 0
[9] 12:10:09.149 SIP: Resolve 816: aaaa udp 127.0.0.1 5060
[9] 12:10:09.149 SIP: Resolve 816: a udp 127.0.0.1 5060
[9] 12:10:09.149 SIP: Resolve 816: udp 127.0.0.1 5060
[5] 12:10:09.149 PACK: SIP Tx udp:127.0.0.1:5060:
SIP/2.0 501 Temporarily Unavailable
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-21bfb19cb37a5db1cbe919bd3d0cb86a;rport=5060
From: "Maxim Shaposhnikov" <sip:3000@pbx.highperf.pro>;tag=203808712
To: <sip:7013@pbx.navica.highperf.pro;user=phone>;tag=98216bdf51
Call-ID: f3b1842e@pbx
CSeq: 28624 INVITE
Contact: <sip:7013@127.0.0.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Vodia-PBX/5.2.0
P-Asserted-Identity: <sip:7013@pbx.highperf.pro>
Content-Length: 0

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We already discussed that.

 

Your suggestion is ridiculos (sorry).

 

There is a big holding company with many departments (subsidiaries). In all versions <= 5.1.1 short numbers calling works fine (inter-domain). I.e. 3000 <-> 7500 - not a problem.

 

What do you suggest? To change numbering? Simply just because you decided to broke this functionality for some reason?

 

Virtually any PBX software (including 3cx, call manager, etc) could easily do that.

 

Please explain, how our customers can continue to use inter-domain calls from SIP phones using short numbering / internal extentions.

 

All suggestions made earlier by you or your colleagues are not working.

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Okay, please open a ticket on that so that we can get this resolved. One of the key questions is if you are using a global trunk or you are using the "try loopback" function in the dial plan.

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OK. As the only way to open a ticket is to pay for the support, I paid 99$ (we are going to buy more licenses anyway) and sent email to "support@vodia.com"

 

Please let me know in case there is any other address to use (or trouble-ticket system).

 

***

 

We are using the "try loopback" function in the dial plan

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