penta s.r.l.

Strangely enough, I've deleted the trunk and recreated it (using the same data) and now it works..

The problem is that I can register to the provider using EyeBeam but not with pbxnsip.

Could you help me to decode the error following this request to register? Thank you. [7] 2009/06/18 23:48:11: SIP Tx udp:83.211.227.21:5060: REGISTER sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 192.168.xx.yy:5060;branch=z9hG4bK-8be9d4687e29492fe66a65a56eaba639;rport From: "111222333" <sip:111222333@voip.eutelia.it>;tag=53223 To: "111222333" <sip:111222333@voip.eutelia.it> Call-ID: bo8pfk8d@pbx CSeq: 47528 REGISTER Max-Forwards: 70 Contact: <sip:111222333@192.168.xx.yy:5060;transport=udp;line=d3d94468>;+sip.instance="<urn:uuid:290191a7-c420-4279-b874-d847a684834c>" User-Agent: pbxnsip-PBX/3.3.0.3165 Authorization: Digest realm="voip.eutelia.it",nonce="4a392a5aea78754b2fea45340545c05306936a85",response="19b7a13e513d4ef2b41c81db67e62c97",username="111222333@83.211.227.21",uri="sip:voip.eutelia.it",qop=auth,nc=00001728,cnonce="49a6eb92",algorithm=MD5 Expires: 3600 Content-Length: 0 [7] 2009/06/18 23:48:11: SIP Rx udp:83.211.227.21:5060: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 192.168.xx.yy:5060;branch=z9hG4bK-8be9d4687e29492fe66a65a56eaba639;rport=5060;received=88.32.xx.yy From: "111222333" <sip:111222333@voip.eutelia.it>;tag=53223 To: "111222333" <sip:111222333@voip.eutelia.it>;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.fcc2 Call-ID: bo8pfk8d@pbx CSeq: 47528 REGISTER Server: SPS EUT GW 01 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=3533 req_src_ip=88.32.xx.yy req_src_port=5060 in_uri=sip:voip.eutelia.it out_uri=sip:voip.eutelia.it via_cnt==1"

3.0.1 release notes say: "For small groups, it is difficult to maintain a constant service in agent groups. Having the phone ringing for a long time gives a bad impression to the caller. In order to cover these problems, the PBX now offers the option to continue playing the music on hold with the mixed IVR messages, even if the agents are ringing already." We've just upgraded our pbxnsip to v3.3 How can we put MoH while our hunt groups are ringing?? Thank you.

I'll check it out, and report. Thank you.

I tried to configure my SPA941/2 phones in order to monitor other extensions, without success. I'm only able to speed dial an extension using the line keys. Is there anyone using these phones with these features on? By the way, in the administration phone page, wich kind of server should I specify? Broadsoft, SPA9000, Asterisk or RFC3265_4235? Phones have last firmware. Nicola

So it's best to wait for 3.1 to be released, before working on the address book. When you think this will happen? I understand.. we were just studying the manual, and the great part of existing examples are US oriented. Anyway, right now we have 2 outbound trunks, one is the Gateway and the other is the VoIP provider. Due to local rates, we are using mainly the PSTN and the VoIP only when we have no more PSTN lines availables. To select the line, we just type '8' or '9' in front of the number.. any tip on this? Can we tell the pbx to use the VoIP trunk when the Gateway is full? Thank you for your answers. Nicola

I've just setup a workaround. The day/night service flag I use is the 79, so I've created a manual service flag 78. In the first auto attendant, pbxnsip evaluates first 78 and then 79. 78 Is normally clear, so 79 does its job most of the times. But if I set 78 manually (using the phone), it jumps to the hunt group that usually answers, skipping the night flag. Nicola

The same happens if the trunk redirects the call to an auto attendant.

Has this issue been solved? It seems I cannot change a service flag from a phone. FYI, the service flag is automatic.

Is anyone using freecall as a VoIP provider? Is it possible to show a caller ID? Thank you.

May be this may seems a quite stupid question, but since it's our first installation, we'll take the risk. :-) We have a dial plan like this: 110 Our_VoIP_Provider 8* 120 Our_PSTN__Gateway 9* In the address book we have to put a 9 in front of the number to route the call to the PSTN Gwy. This works, but callers won't be recognized when calling inbound, for the numbers do not clash. The question arises since I've seen all dial plans examples less or more like our. How do you conciliate address book numbers and dial plans? Do we have to add another line in the Dial Plan for calls originating from speed dial codes? e.g. 110 Our_VoIP_Provider 8* 120 Our_PSTN__Gateway 9* 130 Our_PSTN__Gateway * Thank you Nicola

We have a trunk that redirects inbound calls to an auto-attendant, then to an hunt group. In this case, we see the Caller-ID but not the corresponding address book name (in the call log page of pbxnsip) If we tell the trunk to redirect incoming calls directly to an extension, then we see the matching name. We are using pbxnsip 3.0.1.3023 (Win32).