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Everything posted by vichy

  1. Ive run into this (mind you still running v3.4) where I cant hit that special menu when I have the same cell phone number programmed in multiple extensions. It doesnt seem to matter if they are separated in different domains or not. The system only seems to like to have unique cell phone numbers for all extensions.
  2. I was able to get it to work for outbound calls (didnt try inbound), however I dont use it much. If memory serves there is no G729 codec support on the android and the only other codec we use is g711 so it doesnt work that well unless the device is connected to wifi. For anyone interested, this is only applicable to Android 2.3 and above (Gingerbread). Settings->Call->Use Internet Calling Cheers
  3. Thanks for the idea. MY problem is that I do not want to interrupt the normal call id in the event a call back is required. This would also affect our CDR's correct?
  4. Hello, I'm wondering if there is a way that I can configure the pbx to playback a prerecorded message to my agents (extensions) upon recieving an inbound call that only my agents can hear? For eg... lets say I have a mixed cue of agents that answer calls coming in from several different DID's. each DID represents a different type of call (DID 1 - sales, DID 2 - support, DID 3 - billing). Is there a way I could inject a message to be played back to my agents letting them know "sales call" or "support call" once they answer the incoming call? Once again the determining factor of whats played back is based on the DID that is called. I hope I am explaining this correctly. My actual use for this is not "support" and "billing" its just the easiest thing I could come up with. Thank you,
  5. Hello, I am still running I noticed a strange behavior today and I am not sure if its a bug or if there is something I am missing. I have an multiple extensions set up as DID's, and are redirecting back out to PSTN using the "fwd all calls" section. If a call comes in to one of my extensions and redirects back out but this call is not answered, I have noticed that the Redirected{$r}, Account{$R} and Answered{$C} fields do not always get populated in my CDR's (format - file:disk). ...and actualy in the call log screen of the web interface, the redirect and durations are missing sporadically as well. I have made several test calls and this appears to only work properly about 1 out of every 4-5 calls. The length of the ring duration doesn't seem to matter either. The result is obviously a bunch of cdrs showing no answer and some showing no answer, redirected to xxx-xxx-xxxx. Makes things very confusing Is there some scenario/setting that I am not thinking of that is causing this behavior or is it just a bug? Thanks
  6. Hello, I'm currently running We use the system to redirect alot of numbers i.e. DID's ring into pbx then get redirected back out to PSTN. We currently use the csv flat file method for writing cdr's. I'm wondering if there is any way to distinguish a redirected call that is busy vs not answered? (i.e. if the final redirect number is busy or not answered, not the DID routing to the PBX) Unless I am missing something it seems like there is no way to tell one call from the other. Thanks in advance if anyone has an answer to this.
  7. Well I guess memory serves you well...This works!!! Thanks very much
  8. I found the CDR format parameters in the old wiki. ...tried writing my CDR's using the $S timestamp as an alternate, but found that the recording $t timestamp is still different from this. If what im trying to accomplish ends up not being possible, do you see this as being a feature that could be added? It would be really helpful if I could match the two up somehow. ...im still open to any other solutions you may be able to suggest. Thanks
  9. Hello, I am trying to match up our recorded calls to our CDR's based on the time the call was made as well as extension#/DID. I am doing this by writing the time and extension number in each wave file (i.e. record location is: recordedcalls/$m/$d/$u/$t-$u.wav) Our CDR's are written using the timestamp {$B}. For some reason, it seems as though the call record flag $t uses the call start time, and the CDR flag $B uses the call end time...or something close to it. Either way, the two time stamps are not the same. Is there some other flag I can use for either my CDR's or record location that will produce the same timestamp? I am also open to any other suggestion you may have as to linking the two items with some other sort of unique id. By the way, I am currently running Win32 Thanks!
  10. Thanks for the link. I should reiterate though that I do not need help in importing via csv. I was letting you know that there are bugs with it. I've been importing both extensions and ivr nodes via csv (about 50 per month) for the last 2 years, and this has always been a problem. I normally import the following: type;dial_plan;alias;first_name;mb_enable;mwi;cfa;email_missed;email_status;code cs Everything except what I mentioned in my first post works just fine. I found the column names from here: http://kiwi.pbxnsip.com/index.php/Access_to_the_Database perhaps these lists are no longer valid?
  11. Hi This has been happening for quite some time...I just havent got around to mentioning it. I am still using v3.4.3201 WIN32 but have always had "glitchyness" when using the CSV upload tool to create extensions. The main issue is, if I upload an extension with multiple account numbers/aliases sometimes the system interprets them as one string. For instance, I may upload my account numbers (;alias; field) as ;8005551234 866551234; and they occasionally get imported as one long string with a space....instead of "8005551234" "8665551234". I then have to go back, open each record...delete the second alias, Save, re-type and save again in order for the two numbers to work properly. In addition to this the following fields do not seem to ever work: ;dialplan; - even though I specify a dial plan...it stays at "Default" ;mb-enable; - Mailbox stays enabled even though I specify "FALSE" ;mwi; - Send MWI stays on despite the vlaue "FALSE" same thing with ;email_missed;email_status; Other then that, I dont use the other fields so I am not sure what else may not be working properly. I just wanted to bring it to your attention since your working through v4 and may not have known this was a problem Thanks
  12. I have customized the logo at the top of each page of the web interface, but am wondering if this logo can be different depending on which user and or domain is logging in? For instance, we currently display our company logo on every page...but we have one new client who wishes their company logo to appear at the top instead of ours (for all users in their particular domain). Is this possible? Thanks
  13. Forgot to chime in here and say Thanks! I find the IVR node works best out of the two. Main reason being, I couldnt figure out away to stop the AA from getting stuck in a loop if somone happens to press a key during the playback. Incase anyone comes across this same thing, I set the IVR Node with the DTMF Matching set to T, the SOAP URI set to action:bye and the timeout set to the duration of my recording (15s in my case). Thanks again
  14. Well...I guess no one else is using this appliance for this type of thing. In case it comes up in the future for anyone...I have had to go back to v to get this functionality. This one alows a custom greeting to still be played when max messages is set to 0....it also just disconnects at the end of the message instead of recording a blank message.
  15. Hi there, I just upgraded to from the previous posted version. We have a domain set up specifically to play number change annoucements (reference of calls) via custom mailbox recordings. i.e. "the number xxx has been changed to yyy". In the previous version, I would simply set the maximum number of messages field to 0 and this would allow me to have my custom greeting still play and then the voicemail system would hang up/not record any messages. Since upgrading to the new version, if my mailbox max messages is set to 0, I get a mailbox full message with no custom greeting. It seems as thought the only way for me to replicate what I had in previous versions is to remove the 0 from max messages and let everyone record silent messages. This is fine, but seems as though I am putting unecessary load on the system. Is there a way to replicate what I had before with the newer versions? In the past we have had quite a few uses for this other then just reference of calls. For instance, weather announcement lines....tele-thon type voting polls...etc. Thanks!
  16. I have a two clients (two different domains) that actually share the same support rep. The support rep is a registered sip (snom) extension in Domain 1, and I need the same phone to be available in Domain 2 for direct dial and AutoAttendant dial by name...etc. The extension numbers can be different, this part doesnt matter. How can I accomplish this? I tried a manual registration using IP address, however the support rep has a dynamic IP so the address changes daily. Is this possible? Thanks
  17. Just wanted to chime in here. There are a few similar posts, but didnt know if I should crowd them with my problems...so heres my problem: I have a similar problem on the hunt group side, as others have mentioned. I have a DID that rings into a hunt group like this: Stage 1 502 503 12seconds Stage 2 504 12seconds Final Stage 8502 (502's mailbox) My problem is that 502 and 503 want thier cell phones to ring as they do when thier extensions are called directly (i.e. call cell phone immediately option). The other thing is that 504 is actually an extension in another (different) domain, or an external number..however you want to look at it. What happens is 502 and 503 ring simultanously for 12 seconds, thier cell phones do not ring. Stage 2 is skipped, and the call ends up in 502's mailbox. I understand that some users dont want hunt group stages to follow any external redirection that may be set up for a particular extension, but for this customer this is exactly what they want to have happen. Perhaps its time to add an option in the hunt group settings to "follow extension redirection settings" or something similiar? If I am missing a best practice that would help solve this, then please advise. Thanks! **Edit** I see from other posts that static registrations will do this, however from a user managment perspective (web/pac, etc) most users change thier cell phone#'s or fwd numbers frequently (US and Canada cell phones), so its a bit of a bear on me to constantly check this and change static registrations.
  18. I updated the FW to 7.1.39, and so far everything is the same. As I mentioned earlier..I just updated to pbx v 3.2 from 3.0 a few days back, so I was using this dial plan, as the auto provisioning wasnt a feature previously. I still havent got the dial plan autoprovisioning working correctly yet. Do you see my dial plan as being a contributor to the problem I am experiencing?
  19. Yes, I am able to get my mailbox using *97. I also just realized that I can complete a call to a 10 digit number if I first press one of the "Shared line" buttons I have set up. This gives me a dial tone, and I can call anywhere. I used to be able to dial direct from the keypad and it would pick a co line for me. Here is the dial plan I am using btw: |^(*[0-9]{2})$|sip:\1@\d|d |^(911)$|sip:\1@\d|d |^(611)$|sip:\1@\d|d |^([2-9]{1}[0-9]{9})$|sip:\1@\d|d |^(1[0-9]{10})$|sip:\1@\d|d Attached is a trace of an unsucessful call Thanks! trap.txt
  20. Tried a factory reset, and the problem still seems to exist. I should add that all my buttons work, as far as dialing out to voicemail or other internal extensions. If I try and dial 3 digit extensions...it works. Any 10 Digit number, internal or external produces the same problem. All the snom 320's are running 7.1.35 (the ones that work, as well as the ones that do not). I recently updated the pbx from 3.0 to (Win32) as well, which did not seem to solve the problem either. I notice there are a few later versions available for the 320's. is 7.1.35 still the recommended version? Any other suggestions?
  21. I have a domain set up with several snom 320 phones registered. I recently encountered an issue where two of the phones/extensions suddenly had no outbound calling...but everything else worked fine. Basically, I still get dial tone ....dial out, and nothing happens. Eventually I recieve an email from the PBX regarding the unconnected call. The call from sip:5194949545@sip01.envoicanada.com;user=phone to sip:5199371702@sip01.envoicanada.com has been disconnected because no media session was establised (source= If I look at the log, I see an INVITE, 100...then 401, ACK...then nothing. Whats interesting is that I can fix this issue by either removing the SIP password from this extension, or by using the extension alias as the "Account" that my phone registers with (i.e. instead of using 5199371707/sip password, I use the alias 555/sip password). My question is, shouldnt my phone be able to register with the pbx using either of the numbers in the "Account number(s)" field? (in my case 5199371707 555) And if not, why does this work for some of the snom 320's but not the others? Thanks!
  22. Hello, I'm trying to figure out a way to create a dial plan that allows 7 digit dialling within my local NPA, and all other LD requiring 11 digit dialling (i.e 1-NPA-NXX-XXXX). btw...my Trunk accepts 10 Digit dialing. I figured I would just create two patterns to accomplish this... first one being: 2xxxxxx|3xxxxxx|4xxxxxx|5xxxxxx|6xxxxxx|7xxxxxx|8xxxxxx|9xxxxxx Replacement: NPA* and the second being : 1xxxxxxxxxx replacement: ???? - what I want to do here is strip the 1 prefix from the second pattern. If I can do this, "i think" this should accomplish what I am after. The question is, can I strip digits from a dial pattern using the replacement field. I figure there is probably a way to do it using ERE format, but I cant seem to find a good resource to figure it out. Thanks in advance!
  23. Thats Great News!! Cant Wait, Thanks.
  24. Hello. I am using the same version, on the same platform...and am experiencing the same issue. I ticked off "yes" to all the record options (I'm assuming this is how you record all calls), yet when a call comes in from PSTN and is redirected back out to the PSTN...There is no recording. Should the PBX record calls in this fashion, or am I missing something? Thanks
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