Re: [asterisk-users] Calls Being Randomly Bridged
On Jan 22, 2008 12:22 PM, Steve Davies [EMAIL PROTECTED] wrote: Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way. Cheers, Steve As a follow up, I just spoke with our UK snom distributor, EFL, and they are discussing this with snom already. It seems that there has been some kind of acknowledgment that the transfer behaviour with multiple inbound calls is not ideal, and that an improved behaviour should be forthcoming in a soon to be released version (no promises on which version though) Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Michael J. Liberatore [EMAIL PROTECTED] writes: I do have queues set up but I would have to setup queues for all calls then, even from other inside the office calls. Cause if I disable call waiting, wouldn't that be the same as saying maximum sip connections to the phone = 1? Call waiting off means that someone who calls the phone while at least one call is ongoing will get the busy tone (or e.g. voice mail, if that's what your dial plan says should happen). You can just send the office-calls into the same queue as the calls from outside. You can also give them their own queue, or give them priority. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On 1/22/08, Michael J. Liberatore [EMAIL PROTECTED] wrote: I do have queues set up but I would have to setup queues for all calls then, even from other inside the office calls. Cause if I disable call waiting, wouldn't that be the same as saying maximum sip connections to the phone = 1? Or is call waiting different on the snom phones? Call-Waiting can be disabled on the handset. With snom phones, this can be set to 3 settings via handset, web interface or provisioning: 1) Enabled with beep (Never use this - it is horrible) 2) Enabled, visual only (Use this if you want CWI) 3) Disabled. The phone will allow 1 outbound call, but will send a busy response if a call is already in progress. Cheers, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way. Cheers, Steve On 1/22/08, Michael J. Liberatore [EMAIL PROTECTED] wrote: Wow thanks so much for this, this is a lot of great info. Hopefully enough to catch snom's attention to. Is it possible for you to try 7.x on one of the phones and see if it corrects the problem? What it comes down to, is that the phone is too complicated to handle multiple calls for non technical users. They have to keep track of way too much, even a techie like us could get mixed up sometimes, especially in a high stress doctors office where there are half of the number of receptionists that are reeally needed. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, January 21, 2008 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged I found this problem sufficiently interesting that I went and had a play with our snom phones in the test lab to try and determine what the behavious is. This is with 6.5.13 phones, and I think the results are somewhat inconsistent, particularly if snom are reporting this behaviour as intended as was suggested elsewhere in this thread... We already disable the Call join on Xfer (2 calls): setting, so that can be taken into account in the descriptions below. 1) Simple unattended transfer. This does what is says on the tin regardless of how many other calls are ringing one the handset. It will transfer the call that is in-hand to the number dialled. Achieved with: Transfer, dial number, Tick 2) Simple attended transfer - One caller on the line. Again, this works fine Achieved with: Hold, dial number, tick, wait for answer, transfer, tick Or: Hold, dial number, tick, wait for answer, Hangup Or: Hold, dial number, tick, wait for answer, Transfer, Tick 3) With multiple inbound calls, the behaviour is less well defined. Here is what I found: Call 1 arrives, answer call. Call 2 arrives Call 3 arrives Press hold, dial destination for transfer of call 1, press Tick. Now there are 2 alternatives. a) Unattended. While the call is still ringing, press transfer, you will be offered a list of calls in the order 1, 3, 2 - This is 100% fine. The default destination is call 1 - The last call we dealt with. b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? 4) How to connect two external callers (as per original email). This is a stretch, but I can see it happening... Answer a call, put it on hold, wait for an answer. Re-select the original caller's line to let them know you are about to transfer their call. Press transfer (another call has come in in the meantime) the list you are offered defaults to the new (unanswered) call, and not the recently dialled and answered transferee. Not good really :( Basically, whatever calls the operator has had DIRECT involvement with should be kept at the top of the stack of calls, so that any default operations relate to those topmost calls. New calls go at the bottom of the stack, and stay there until there is some direct interraction with them. How hard is that? Just my 2p. Steve -Original Message- Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me
Re: [asterisk-users] Calls Being Randomly Bridged
Hi Mike, For starters disable Call join on Xfer (2 calls): on the phones. Since the setup has 6.2.x, it most likely doesn't have the setting Allow incoming calls redirection through programmable keys available on 7.1.30 for snom360. You might wanna try this version on a test system and see if it helps in that environment. The problem, as discussed, seems to be originating when calls are parked on orbits that are mixing the two calls together. As long as you are debugging the issue, you should probably ask your friend to disable this practice and have a look at the call parking mechanism. Regards, Usman. - Usman Tahir snom technology AG www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - -Original Message- Message: 11 Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Hi, My personal experience of this is that the call transfer facility on older vesions of snoms (6.2.x is rather old now) is quite hard to get to grips with - Particularly when managing multiple calls. Newer versions seem to be better, but generally you need to train people to look at the screen and use the silver keypad to choose the call to transfer to. The worst situation is where 2 calls come in with no caller-id, so you have no clue which call to transfer, and the phone does not store sufficient state to automatically transfer the last call I was on to the current call I am on, or even make this the default transfer target, which is going to be the requirement 99% of the time... We use 6.5.12 firmware and upwards to 6.5.15. We have an open support ticket on 7.1.30 causing calls to hangup when put on hold, so are not brave enough to go there yet. Regards, Steve On 1/21/08, Usman Tahir [EMAIL PROTECTED] wrote: Hi Mike, For starters disable Call join on Xfer (2 calls): on the phones. Since the setup has 6.2.x, it most likely doesn't have the setting Allow incoming calls redirection through programmable keys available on 7.1.30for snom360. You might wanna try this version on a test system and see if it helps in that environment. The problem, as discussed, seems to be originating when calls are parked on orbits that are mixing the two calls together. As long as you are debugging the issue, you should probably ask your friend to disable this practice and have a look at the call parking mechanism. Regards, Usman. - -Original Message- Message: 11 Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
I found this problem sufficiently interesting that I went and had a play with our snom phones in the test lab to try and determine what the behavious is. This is with 6.5.13 phones, and I think the results are somewhat inconsistent, particularly if snom are reporting this behaviour as intended as was suggested elsewhere in this thread... We already disable the Call join on Xfer (2 calls): setting, so that can be taken into account in the descriptions below. 1) Simple unattended transfer. This does what is says on the tin regardless of how many other calls are ringing one the handset. It will transfer the call that is in-hand to the number dialled. Achieved with: Transfer, dial number, Tick 2) Simple attended transfer - One caller on the line. Again, this works fine Achieved with: Hold, dial number, tick, wait for answer, transfer, tick Or: Hold, dial number, tick, wait for answer, Hangup Or: Hold, dial number, tick, wait for answer, Transfer, Tick 3) With multiple inbound calls, the behaviour is less well defined. Here is what I found: Call 1 arrives, answer call. Call 2 arrives Call 3 arrives Press hold, dial destination for transfer of call 1, press Tick. Now there are 2 alternatives. a) Unattended. While the call is still ringing, press transfer, you will be offered a list of calls in the order 1, 3, 2 - This is 100% fine. The default destination is call 1 - The last call we dealt with. b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? 4) How to connect two external callers (as per original email). This is a stretch, but I can see it happening... Answer a call, put it on hold, wait for an answer. Re-select the original caller's line to let them know you are about to transfer their call. Press transfer (another call has come in in the meantime) the list you are offered defaults to the new (unanswered) call, and not the recently dialled and answered transferee. Not good really :( Basically, whatever calls the operator has had DIRECT involvement with should be kept at the top of the stack of calls, so that any default operations relate to those topmost calls. New calls go at the bottom of the stack, and stay there until there is some direct interraction with them. How hard is that? Just my 2p. Steve -Original Message- Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Oh, and the workaround is to disable call-waiting on the snom phone, and use a queue to hold callers if the line is busy. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Steve Davies [EMAIL PROTECTED] writes: Oh, and the workaround is to disable call-waiting on the snom phone, and use a queue to hold callers if the line is busy. Isn't that pretty much the only way, even if the Snom bugs are fixed? Getting the buzz from call waiting every 30 seconds must be quite stressful. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On 20/01/2008, Michael J. Liberatore [EMAIL PROTECTED] wrote: They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. In the SNOM settings there are two options that you should set to No. That is Call Join on Hangup and Xfer on Hangup. (Or names similar to that). Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On 21/01/2008, Steve Davies [EMAIL PROTECTED] wrote: b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? Yes - I completely agree that the SNOM attended-transfer is screwy in the presence of a third call. It causes problems if you have a long-running call and want to leave that on hold whilst handling another call that came in, or if a third call starts to ring in the middle of transferring a pre-existing call. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Yes these 2 options have been set to NO all along. I double checked too. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Monday, January 21, 2008 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged On 20/01/2008, Michael J. Liberatore [EMAIL PROTECTED] wrote: They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. In the SNOM settings there are two options that you should set to No. That is Call Join on Hangup and Xfer on Hangup. (Or names similar to that). Steve This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
I do have queues set up but I would have to setup queues for all calls then, even from other inside the office calls. Cause if I disable call waiting, wouldn't that be the same as saying maximum sip connections to the phone = 1? Or is call waiting different on the snom phones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, January 21, 2008 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged Oh, and the workaround is to disable call-waiting on the snom phone, and use a queue to hold callers if the line is busy. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Wow thanks so much for this, this is a lot of great info. Hopefully enough to catch snom's attention to. Is it possible for you to try 7.x on one of the phones and see if it corrects the problem? What it comes down to, is that the phone is too complicated to handle multiple calls for non technical users. They have to keep track of way too much, even a techie like us could get mixed up sometimes, especially in a high stress doctors office where there are half of the number of receptionists that are reeally needed. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Monday, January 21, 2008 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged I found this problem sufficiently interesting that I went and had a play with our snom phones in the test lab to try and determine what the behavious is. This is with 6.5.13 phones, and I think the results are somewhat inconsistent, particularly if snom are reporting this behaviour as intended as was suggested elsewhere in this thread... We already disable the Call join on Xfer (2 calls): setting, so that can be taken into account in the descriptions below. 1) Simple unattended transfer. This does what is says on the tin regardless of how many other calls are ringing one the handset. It will transfer the call that is in-hand to the number dialled. Achieved with: Transfer, dial number, Tick 2) Simple attended transfer - One caller on the line. Again, this works fine Achieved with: Hold, dial number, tick, wait for answer, transfer, tick Or: Hold, dial number, tick, wait for answer, Hangup Or: Hold, dial number, tick, wait for answer, Transfer, Tick 3) With multiple inbound calls, the behaviour is less well defined. Here is what I found: Call 1 arrives, answer call. Call 2 arrives Call 3 arrives Press hold, dial destination for transfer of call 1, press Tick. Now there are 2 alternatives. a) Unattended. While the call is still ringing, press transfer, you will be offered a list of calls in the order 1, 3, 2 - This is 100% fine. The default destination is call 1 - The last call we dealt with. b) Attended. Wait for the call to answer, Press transfer, you will be ordered a list of calls in the order 3, 2, 1 - This is 100% wrong. The call you want is LAST in the list. If you have no CID, or have forgotten the CID of the caller, you cannot easily transfer the right call, and might instead connect the wrong caller. Why would you offer an unanswered call over an answered one anyway??? 4) How to connect two external callers (as per original email). This is a stretch, but I can see it happening... Answer a call, put it on hold, wait for an answer. Re-select the original caller's line to let them know you are about to transfer their call. Press transfer (another call has come in in the meantime) the list you are offered defaults to the new (unanswered) call, and not the recently dialled and answered transferee. Not good really :( Basically, whatever calls the operator has had DIRECT involvement with should be kept at the top of the stack of calls, so that any default operations relate to those topmost calls. New calls go at the bottom of the stack, and stay there until there is some direct interraction with them. How hard is that? Just my 2p. Steve -Original Message- Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike ___ -- Bandwidth and Colocation Provided by http://www.api
Re: [asterisk-users] Calls Being Randomly Bridged
Tilghman Lesher schreef: On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. I had exact the same problem in using the snom 360, it's too easy to bridge 2 calls, it isn't a bug, it works as designed but transfering a call on a 360 isn't as user friendly as it should be, specially when many calls are incoming. I've replaced the snom 360 by a linksys 962 and disabled blind transfer. But be warned. When using the 962 and the extra panel train you users using the numeric keypad when transfering calls, using the extra buttonpanel when transferring calls randomly results in loosing calls. Personally i'am still looking for a good station when a lot of incoming trafic is on a main station. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fons van der Beek Sent: Sunday, January 20, 2008 3:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged Tilghman Lesher schreef: On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. I had exact the same problem in using the snom 360, it's too easy to bridge 2 calls, it isn't a bug, it works as designed but transfering a call on a 360 isn't as user friendly as it should be, specially when many calls are incoming. I've replaced the snom 360 by a linksys 962 and disabled blind transfer. But be warned. When using the 962 and the extra panel train you users using the numeric keypad when transfering calls, using the extra buttonpanel when transferring calls randomly results in loosing calls. Personally i'am still looking for a good station when a lot of incoming trafic is on a main station. I think this is the cause too. I checked the logs for parking to direct spots and I didn't see any of that going on so I think this is the likely cause. I disabled the conference button but I think the problem is with transfers as you mentioned. Can anyone think of a way to prevent connecting two callers with the transfer function? Either in the phone or asterisk? I need to have the ability to transfer, but NEVER connect two incoming callers, only connect an incoming caller with a different internal phone. How do you think 2 outside callers are getting bridged with transfering? Thanks Mike Also to the person asking for more detail logs, I will try to get them, they can never tell me exactly when this happens only that it happened a bunch of times this week This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Michael J. Liberatore schreef: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fons van der Beek Sent: Sunday, January 20, 2008 3:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged Tilghman Lesher schreef: On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. I had exact the same problem in using the snom 360, it's too easy to bridge 2 calls, it isn't a bug, it works as designed but transfering a call on a 360 isn't as user friendly as it should be, specially when many calls are incoming. I've replaced the snom 360 by a linksys 962 and disabled blind transfer. But be warned. When using the 962 and the extra panel train you users using the numeric keypad when transfering calls, using the extra buttonpanel when transferring calls randomly results in loosing calls. Personally i'am still looking for a good station when a lot of incoming trafic is on a main station. I think this is the cause too. I checked the logs for parking to direct spots and I didn't see any of that going on so I think this is the likely cause. I disabled the conference button but I think the problem is with transfers as you mentioned. Can anyone think of a way to prevent connecting two callers with the transfer function? Either in the phone or asterisk? I need to have the ability to transfer, but NEVER connect two incoming callers, only connect an incoming caller with a different internal phone. How do you think 2 outside callers are getting bridged with transfering? Thanks Mike Also to the person asking for more detail logs, I will try to get them, they can never tell me exactly when this happens only that it happened a bunch of times this week On the snom 360 If you pay close attention when you transfer the calls, you can see the names/numbers of the calling partners by using the cursor button (the round button with arrows) you can select to who you want to transfer to. It's an user issue, but you can't blame the user when there is a lot incoming traffic it takes too many button presses and careful attention to make a correct transfer. How to disable it? I don't know but i faced the problem that users occasionally want to bridge calls. e.g. someone calls for a person that only can be reached by Cellphone, this can be accomplished by asterisk and is often needed. Personally I'm still looking for a good solution for a central station that is easy to use and has a professional appeal, i thought the linksys 962+932 was it, but it has also some drawbacks. One(or two) button attended transfer is not reliable. certainly not when there are 2 or three simultaneously incoming calls. It gets confusing at that time. If anyone has any suggestions don't hesitate to make them! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] Calls Being Randomly Bridged
On 13:06, Sun 20 Jan 08, Fons van der Beek wrote: Michael J. Liberatore schreef: On the snom 360 If you pay close attention when you transfer the calls, you can see the names/numbers of the calling partners by using the cursor button (the round button with arrows) you can select to who you want to transfer to. It's an user issue, but you can't blame the user when there is a lot incoming traffic it takes too many button presses and careful attention to make a correct transfer. How to disable it? I don't know but i faced the problem that users occasionally want to bridge calls. e.g. someone calls for a person that only can be reached by Cellphone, this can be accomplished by asterisk and is often needed. Personally I'm still looking for a good solution for a central station that is easy to use and has a professional appeal, i thought the linksys 962+932 was it, but it has also some drawbacks. One(or two) button attended transfer is not reliable. certainly not when there are 2 or three simultaneously incoming calls. It gets confusing at that time. If anyone has any suggestions don't hesitate to make them! We noticed the same problem. We tracked it down to this: snom gets a call and answers it. snom talks to the user. While talking to the user a second call comes in (callwaiting is enabled) user wants to be transferred so the snom operator hits the transfer button. snom automagically selects the second incoming call as target and bridges them. We called snom and they told us it's by design. We have not tested the new 7.1.30 firmware, but there have been a lot of changes in the hold/transfer/fwd functions, so maybe they fixed it. We replaced the phones by aastra's on this particular location and everything is fine now. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, January 20, 2008 7:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Calls Being Randomly Bridged On 13:06, Sun 20 Jan 08, Fons van der Beek wrote: Michael J. Liberatore schreef: On the snom 360 If you pay close attention when you transfer the calls, you can see the names/numbers of the calling partners by using the cursor button (the round button with arrows) you can select to who you want to transfer to. It's an user issue, but you can't blame the user when there is a lot incoming traffic it takes too many button presses and careful attention to make a correct transfer. How to disable it? I don't know but i faced the problem that users occasionally want to bridge calls. e.g. someone calls for a person that only can be reached by Cellphone, this can be accomplished by asterisk and is often needed. Personally I'm still looking for a good solution for a central station that is easy to use and has a professional appeal, i thought the linksys 962+932 was it, but it has also some drawbacks. One(or two) button attended transfer is not reliable. certainly not when there are 2 or three simultaneously incoming calls. It gets confusing at that time. If anyone has any suggestions don't hesitate to make them! We noticed the same problem. .We tracked it down to this: snom gets a call and answers it. snom talks to the user. While talking to the user a second call comes in (callwaiting is enabled) user wants to be transferred so the snom operator hits the transfer button. snom automagically selects the second incoming call as target and bridges them. We called snom and they told us it's by design. We have not tested the new 7.1.30 firmware, but there have been a lot of changes in the hold/transfer/fwd functions, so maybe they fixed it. We replaced the phones by aastra's on this particular location and everything is fine now. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Thanks for the info, anyone else think this is CRAZY!!?? To assume that you want to bridge the 2 calls when you press transfer is crazy. I am on the phone with patient, another call comes in, I want to transfer call to another receptionist so I can handle the new call, and when I hit transfer it bridges the 2 incoming calls? Does anyone else see the dumbness to this? 99% of the time you wouldn't want them bridged, so having it as a default feature by design that cant be changedseems nuts. Unless I am understanding what you are saying wrong. I am def. gonna try the new 7.x firmware just released and hope it fixed the problem. It's a shame cause snom's could be great phones but the firmware has always sucked. The new polycoms look nice but they don't have the line buttons like snom does, I need to have the blf buttons with lights for like 3 or 4 lines, and then the other extensions with blf enabled. The polycom's don't have this, only on the screen which non tech users HATE. Aastra I tried once and I think it had the blf buttons but not as many as snom and I had trouble with the firmware, I don't remember which model. I have a couple linkssy sphones, they are nice but again missing the blf/line buttons so do cisco's. Does anyone like cisco with asterisk? I would assume if you get the sip firmware that they are quite reliable, since lots of large corp's use them. But they have similar issues with no blf/line buttons. The granstream gxp-2000 has the blf/line buttons but they are terrible phones. Am I missing any phones? Any other suggestions? How do you get around the no blf/line buttons on polycom and linksys? No tech users hate it. Anyone use the new polycoms? They seem nice. Now going back to the issue, I will never need to bridge 2 outside calls, is there a way to disable it in asterisk some how? Never let 2 outside callers get bridged? Maybe in configs or code? Thanks Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Calls Being Randomly Bridged
The granstream gxp-2000 has the blf/line buttons but they are terrible phones. Am I missing any phones? Any other suggestions? I have to agree with your point - the transfer on the Snom's is not good if you have to juggle several calls. The Polycom transfer system is probably the best, but a Polycom plus a sidecar is lot more money than a Snom with it's built in 12 buttons. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
2008/1/21, Michael J. Liberatore [EMAIL PROTECTED]: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, January 20, 2008 7:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Calls Being Randomly Bridged On 13:06, Sun 20 Jan 08, Fons van der Beek wrote: Michael J. Liberatore schreef: On the snom 360 If you pay close attention when you transfer the calls, you can see the names/numbers of the calling partners by using the cursor button snip Does anyone like cisco with asterisk? I would assume if you get the sip firmware that they are quite reliable, since lots of large corp's use them. But they have similar issues with no blf/line buttons. To my knowledge, trouble with Cisco SIP phones is you can't localize them with Asterisk : menu are displayed in english. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On Sat, Jan 19, 2008 at 09:32:42PM -0500, Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. Can you provide a more detailed trace of such an event? (Use more verbose logging, and such) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
