John

Hello, we are facing the following issue with 6 Linksys SPA 962 devices. About 1 in 5 calls, whether incoming, outgoing or even between extensions have no audio at all. The phone of the called party rings (so the signaling is OK, even though the caller cannot hear the ringing tone while the phone of the called party is ringing) but when the call is answered, both parties hear nothing. The problem is caused by the phones, because I replaced one with a yealink device and the yealink worked without issues. I have spent a lots of time so far experimenting with various settings without success. Has anyone encountered a similar issue? Do you have any ideas or suggestions? Thank you very much in advance, John

The "Use same settings for all domains" is already set to yes. I am pretty sure that this is a PBX issue. The system does not even try to send e-mails most of the time. In addition, many applications and services use our exchange server to send notifications, alarms etc. The e-mails from all systems with the exemption of the PBX are sent successfully. Before upgrading to 5.2.3 the PBX could also send e-mails without issues.

Also the "Administratively log in these agents: " and "Administratively log out these agents:" fields aren't available in the edit Agent Group screen anymore. Did you move them (where) or remove them?

Hello, Software-Version: 5.2.3 (Debian64) Build Date: Jun 28 2014 06:40:10 we upgraded to 5.2.3 after the bug with the e-mail had been fixed. But the PBX cannot send any e-mails. The system does not even try to send an e-mail after i.e. an unanswered call or after a voicemail was left (nothing appears in the logfile). When we try to send a test message (from the e-mail setup screen), the following error appears in the logfile: [3] 14:37:43.739 EMAI: Last message repeated 2 times [7] 14:37:43.739 EMAI: XXX.XXX.XXX.XXX: Connect to XXX.XXX.XXX.XXX:25 [8] 14:37:43.743 EMAI: Received SMTP traffic 220 mailsrv01.XXXXXXXX.XX Microsoft ESMTP MAIL Service ready at Mon, 7 Jul 2014 14:38:53 +0300 [8] 14:37:43.743 EMAI: Send SMTP traffic EHLO localhost [8] 14:37:43.745 EMAI: Received SMTP traffic 250-mailsrv01.remoteaXXXgr Hello [XX.XXX.XXX.XX] 250-SIZE 10485760 250-PIPELINING 250-DSN 250-ENHANCEDSTATUSCODES 250-X-ANONYMOUSTLS 250-AUTH NTLM LOGIN 250-X-EXPS GSSAPI NTLM 250-8BITMIME 250-BINARYMIME 250-CHUNKING 250-XEXCH50 250-XRDST 250 XSHADOW [8] 14:37:43.745 EMAI: Send SMTP traffic AUTH LOGIN [8] 14:37:43.747 EMAI: Received SMTP traffic 334 VXNlcm5hbWU6 [5] 14:37:43.747 EMAI: SMTP Server returned unexpected authentication result 334 [8] 14:37:43.747 EMAI: Send SMTP traffic QUIT [8] 14:37:43.748 EMAI: Received SMTP traffic 334 UGFzc3dvcmQ6 The mail server works fine. According to the server administrator, the pbx does not send the auth login to the server. Cannot you please check it and possibly upload a new Debian build? This is urgent.

@ voipguy thank you for your reply but I don't think that this is the cause in our case. I checked the logs without finding anything regarding a date/time change. In addition, in our case no call was dropped. And we never perform administrative tasks while the system is in use. We always send maintenance notifications to our customers in advance.

Hello, (Vodia v5.2.2 hosted, Debian OS) I noticed that the call duration reported in the call history for some calls (very few for the time being) is wrong. There isn't a pattern on the type of calls. Some are incoming calls, others are outgoing. Also calls to other extensions in the same domain and interdomain calls (with loopback enabled, interdomain calls aren't sent to the sip proxy). Even unanswered calls. There isn't also a pattern on the time of the day. For instance: The first one is a call I placed from my mobile to one of my colleagues. We talked for about 10 minutes. The duration of the call below is also wrong (besides, the maximum call duration is set globally to 4 hours). Is there a bug or something else?

This helps a lot, thank you very much.

How can I do that? Is "rec" a file attribute? ls -la returns something like this: -rw-r--r-- 1 root staff 10200 May 19 15:18 msg447.wav -rw-r--r-- 1 root staff 18326 May 20 11:31 msg451.wav -rw-r--r-- 1 root staff 5000 May 20 13:45 msg453.wav -rw-r--r-- 1 root staff 3636 May 20 13:48 msg455.wav -rw-r--r-- 1 root staff 14686 May 20 13:49 msg457.wav -rw-r--r-- 1 root staff 2986 May 20 16:21 msg461.wav while the file command returns something like this: john@snom1:/usr/local/snomONE/recordings$ file msg166.wav msg166.wav: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz I managed to do something similar with a Yealink phone. There is a Record Soft Key which doesn't even need a value (*93, *94) to start and stop the recording. And the tones aren't audible.

Thank you for the response, An ad-hoc recording issue. We have a customer who wants the users in his domain to record only specific calls using the *93 star code. I recommended to enable automatic recordings but he doesn't want to and he is explicit about that. The files of ad-hoc recordings are all saved in the same directory (the recordings directory in the snomONE working directory), regardless of the domain or the user who records the call, along with voicemails etc. Moreover, the filename format for the ad-hoc recordings is msg+recording_number.wav. We host about 20 customer domains in this server (and we are going to add more). The customer mentioned above wants us to send him the ad-hoc recordings from his users. But with all ad-hoc recordings saved in the same folder and with that filename format, I cannot write a script to move his recordings only, for instance to his FTP. The only way to identify his files is to hear them. All. This can't be done obviously. By the way, when I login as system admin I cannot see the ad-hoc recordings, I must login as the user who recorded them. But I think this is the intended behavior. If not, then it's a bug. Not sure if I can see them If I login as a domain admin, I haven't tried it yet. Finally, something else: by using the *93 and *94 star codes to start and stop a recording, the recording tones are audible. If we buy a snom phone with a record button will the tones also be audible (ideally we don't want to).

Hello, is there a way to create a folder hierarchy and adjust the filename for the manual recordings, just like the automatic recordings? Now all manual recordings from all users in all domains are stored in the same folder (/usr/local/snomONE/recordings) along with other recordings (for instance, voicemail messages). In multidomain environments this is not working because there is no way to find by which customer and user each message was recorded without hearing it.

Yes, something like that. But for the Camp On to work, I need to disable the call waiting in all the devices, which the users do not want.

Hello, a secretary of one of our customers has requested the following feature: When she worked for another company, many coworkers were asking her to notify them when the boss's line was available (=not busy) to call him. She had a secretary console attached to her phone where she could check the status of all extensions (available, ringing or busy). If an extension was busy, she could press the associated button and the pbx displayed a message that the extension was busy at the moment, followed by a prompt to call her when the call ended. That way she could concentrate to her job without having to check herself every few minutes if the extension was finally available. Now her colleagues have similar requests and she asked if we can implement something similar. AFAIK the pbx doesn't have a feature like that. Is it possible that it can be done through the phone itself? She has a Yealink phone (SIP T28P) with a console attached (EXP 38). I read the manual and searched the internet but I found nothing. Thank you very much in advance for any suggestions/ ideas.

Maybe I didn't express it correctly. We cannot see some extensions at all in the WAC, not just the changes in their status. The domain includes the extensions 129, 141, 142, 145, 148, 149, 150, 155, 156, 157, 159, 161, 162, 163, 164, 165, 166, 167, 182, 183, 184, 185, 186, 187, 188 and 189. Here is what I see when I log in to my WAC:

Hello, hosted (debian based) v5.2.1 version here. It seems there is an issue with the WAC: it works but only the status of some extensions (about 50%, always the same) is displayed. All users use Yealink voip phones and some of them also use sip clients (mostly Bria) in their mobile phones. Do you have any ideas why is this happening? The settings are the same for all extensions. Thank you very much in advance, John

Hello, I am a collegue of Nick. We want those kind of calls to be handled by the Snomone only (with the "loopback detection" option turned off). We don't want the calls to be routed to a sip proxy.. Is this possible? Can you please suggest a configuration (for instance a new dial plan)? To be more specific: extension 120 from the domain 112310.z3.vpbx.gr is trying to call the DID +302109558300. This DID is assigned to extension 340, which is at the domain 115396.z3.vpbx.gr. Can you please suggest a dial plan rule for this interdomain call? John.

Looking at the tcp dump it seems you're right. The user says otherwise (he says that the transfer completed and the second extension was able to talk for a few seconds). I'll wait for another incident and a new trace to look at it again.

Thank you very much for your response. Could you please be more specific. What would you suggest us to check? As for the version, we haven't yet tested version 4 thoroughly. Do you think that the problem can be solved by upgrading?

Hello, I would like to describe you a serious problem that we have with our PBXnSIP installation (3.4.0.3201hosted pro plus - Linux version) and kindly ask you for your quick response. Very often calls are dropped during the conversation. This happen both in external calls and in internal calls (calls between pbxnsip extensions). I'm attaching a tcpdump in order to help you to find the reason. Please respond as soon as possible providing us the solution. Best Regards, Michael Aslanoglou Dropped_Calls.zip

Hi, We operate a multidomain environment. We also use a global trunk for all domains. I would like to ask what the correct parameters in the Replacement Field of a Dial Plan are, in order the PBX to send the information below to the ITSP: 1. The caller’s (extension) domain AND/OR 2. The system’s internal user ID I have created the following rule: Pattern: ^(00|\+)([1-9][0-9]*) --> Replacement: sip:\2@\r;user=phone When I call 00441133508795 for example, the information below is included to the INVITE that the system sends to the ITSP: “… To: <sip:441133508795@62.205.34.19;user=phone> …”, where 62.205.34.19 is the ITSP’s gateway IP address. Instead, for billing related issues, we need the pbx to send: “… To: <sip:441133508795@mydomain.com;user=phone> …”, where mydomain.com is the caller’s domain OR “… To: <sip:441133508795@7175;user=phone> …”, where 7175 is the extension’s system internal ID (not the extension name) in the PBX. Thank you very much in advance.

Hi, we have the following problem with the programmable keys (DSS Keys) in Yealink T26 and T28 phones every time we try to make an atended transfer call: Senario A 1. User A calls User B, 2. To make and attended transfer to user C, user B has to press the transfer key and dial his extension number. User B and user C can talk to each other and user B can complete the call transfer by pressing the transfer key again. Senario B When user B presses the transfer key and a pre-programmed function key (instead of dialing the extension number of the C user), he cannot talk with user C. If user B press the transfer key again, the transfer will be completed succesfully but it's like a blind transfer that way. A customer tested the attended call transfer with another IP PBX and it worked without problems both by dialing the extension number and by pressing the function key. PBX Version: Version: 3.4.0.3201 Hosted Pro Plus Operating System: CentOS Linux. We have upgraded the firmware in all phones to the latest version. Do you know how we can solve that problem?

Hi, I need some help with the following scenario: I am trying to connect one pbxnsip pbx with a nortel pbx (the nortel pbx is behind a gateway). I have set up a trunk with account name 113564 (trunk type: gateway, nothing in the "Trunk ANI:" field). The problem is that when an extension on the pbxnxip pbx calls an extension on the nortel pbx, the account name of the trunk is desplayed (neither the caller's extension number nor the ANI) . As a result, the callee can't know who is calling without answering the call. Do you know what I have to change so that a the caller's name / extension number/ ANI is displayed (so that the caller could be identified)? Thanks in advance, John

Hi, is it possible to add a "cli range" field so that each customer (domain) can choose clis (as DDIs and/ or as ANIs) from that range only? This (for security reasons) is a required feature for hosted environments. Thanks, John

Thank you for your response. Didnt't work for me. After adding # the system waits forever for the extension number. So the IVR options (e.g. "For sales press 1 ...) aren't working. Does a solution exists? Thanks, John

Hi, I have 2 questions about the auto attendant feature. 1. Is there any way to set the delay between the first announcement ("welcome to ...") and the next one? Currently the delay between the announcements is about 5 seconds and I think it's long. Please note that I switched off the prompt for the Extension number through the Edit Auto Attendant page. 2. Our first message in the Direct Destinations list is "For sales press 1" with a Sales Hunt Group as the Destination. Extension numbers (3 digits long) also start with 1. When an outside caller tries to enter an extension number (starting with 1) the system always forwards the call to the sales hunt group, ignoring the extension number. Do you have any idea why this is happening? Thanks, John

Thank you for your response, I have a PBX with multiple domains. In this case how can I configure VoIP phones or soft phones that do not have a "domain" field (see an example below). Please note that the users can access the PBX only by using the IP Address of the PBX server.