Vernon

Hello, Was wondering if you could provide any insights on how to resolve no audio on calls made via the user portal. I tried following all the necessary steps, i have a domain certificate, using chrome, and i have logged in via HTTPS. I'm able to make the call go out and i'm able to put the call on hold with the MoH kicking in, but no audio between users. Currently using v59.8 on Windows. Any insights would be helpful, thank you.

Yeah i meant GAPS with the Snom PBX not with the phones . I had the same scenario of having to add the MAC's into the GAPS beforehand. Thank you for taking the time to respond, i guess you and i had similar experiences with it. I'll be on the lookout for when the special feature hits.

Has anyone had luck integrating the GAPS with Snom? I was trying to integrate it via the Phones -> Settings -> Provisioning Server Redirection section with the username/password for the account. Tried inputting a MAC for one of the grandstream phones on a test extension but it didn't seem to create any records on GAPS. I was testing this on the 56.0 release of Vodia.

Hello, Thanks for replying. I do see agents sometimes appearing during the general recorded calls section but that seems to occur only if someone dials an agent (extension) directly. In regards to the Agent Group it never seems to show up even though that information is technically present during the call log. See below for example. Agent Group Call Log: Time From To Agent Waiting Time Ring Time Talk Time (Hold Time) 9/21 1:53P "Anonymous" <sip:anonymous@domain.domain.com> "Center" <sip:Telephone#@domain.domain.com> "Jaime" <sip:224@domain.domain.com> 00:15 00:06 02:20 00:44 Agent Group Recorded Calls: Start From To Agent Play 9/21/2016, 2:10:20 PM Anonymous (anonymous) Center (Telephone#) ## Sorry for the poor formatting i simply copied and edited the data. ## indicates no data. So there are multiple agent groups and all the agents are logged into them. The call flow is designed so that all calls go through the agent groups. I'm looking for guidance on how to have the Recorded Call section properly show the Agent/Extension involved in the call as the PBX clearly has this data but is not displaying it in other sections. Thank you

Hello, I'm looking at the agent group account and noticed that under Recorded Call section there are the following outputs: Start From To Agent Play Everything but the Agent is being displayed in the CDR. When i go into the Call Log i can see that there's data for every field but the recorded call section doesn't display the agent involved in that call. My question is, what would i need to modify for the Recorded Call section to display the agent? PBX is version 5.5.0 Thank you.

Update: I have been playing around with some of the firmware's and decided to use the patch version 8.7.3.5.29. It seemed to have fixed the phone from taking an extra CO line for an incoming call from an attendant. Just out of testing sake i did put back the latest firmware and the same issue came out. So it would seem to be a firmware related nuisance. Hope this helps to whoever else runs into the same issue.

Hello, I noticed a rather weird issue with the Snom 720 and the CO lines when an auto-attendant is involved. The scenario goes like this: A call comes in and hits the attendant. On all the phones the first line lights up as they await for the caller to choose the option on the AA. Once a call is made to say ext 221 (Snom 720) the user of the phone now sees two co lines being used, but this is still the only call on the system. This type of scenario doesn't occur when a Snom 320 is involved, only the first line gets lit up during the entire call flow. Is there a feature or setting that's causing the Snom 720 to use two buttons instead of one for a single incoming call? I was able to circumvent the system from taking the second CO line button by creating a private line but unfortunately that is one too many lights for some individuals. Edit: Forgot to mention the firmwares. Snom 720: 8.7.5.35 Snom 320: 8.7.5.35 PBX: 5.3.2.a Thanks for your time.

Hello, I'm wondering if the SIP ip replacement list can accommodate dynamic IP's. It worked like a charm when using static IP for replacement but the customer has a failover connection for the secondary internet that's not on a static IP. So when the fail over kicks in we get one way audio. So my question if it's possible to do the sip replacement with a dynamic IP. I looked at the wiki entry (http://wiki.snomone.com/index.php?title=SIP_Ports) and it doesn't mention this and i have a feeling it's not possible but it doesn't hurt to ask. Running the Vodia PBX on 5.4.0. Thanks for your time.

Hello, I need some help setting up inter-domain trunks/dial plans. I tried going over the wiki and there is a big wealth of information but i still need a little bit of help. Here is my scenario. I have a PBX with two domains. Domain A and domain B. I want to be able to dial within these two domains interchangeably by using prefixes. So for example to reach Domain B from Domain A use prefix 7. To reach domain A from B use prefix 9. Some of the extensions are overlapping so hence the prefix, i was trying to use the pattern from this wiki page: http://wiki.vodia.com/index.php?title=Dial_Plan_Inter-domain But i noticed that wiki is based on the 4.5 PBX and version 5 uses a different method. Which is here: http://wiki.vodia.com/index.php?title=Vodia_PBX_V5_inter-domain_dialing Is it possible to recreate the 4.5 scenario by using the prefixes? Creating a +10 digit for each domain extension as outlined in the wiki is rather tiresome when having to deal with 100+ extensions. How would the dial plan/trunk setup look like? Based on the wiki the recommendation is to create a dummy PBX with the global trunk and then use country/10 digit codes that searches for the correct PBX. The only work-around to what i've found so far is to create a unique IVR/calling card node and use it as a means of dialing through the respective domains. Would you be able to provide any insight on how to create the inter-domain dialing with prefixes? Is that even possible on version 5? Current PBX is 5.3.2

I noticed in the starcode section 00 to 60 is reserved for speed dials, but they don't have any setup on the PBX or on the phones. They do have a park orbit setup, however so do other customers but these starcodes don't show up in their PBX call logs. These starcodes also show up on the phone's web interface as a dialed call. Why would a phone dial interbal PBX codes? Is there a specific scenario that generates these codes other than picking up specific calls?

Hello, I have run into an odd issue. Our customer has about 13 Snom 320's all running firmware version 8.4.35 and they are experiencing a lot of dropped calls on incoming and outgoing calls. The PBX is on a 5.2.5 CentOS system. So far this is the only domain on this server experiencing these issues. I've worked to replace nearly every aspect of the network save for the phones and the issue is still occurring. The only thing i've noticed in the logs is that the phones occasionally send random *60xxxx codes that don't make sense. I've checked with the customer and they are not dialing these *60's by themselves and appear in both PBX's call log and phone log. I was hoping you could give me any insight as to what the issue is, i believe that if i can solve the cause of these *60 occurring then it would resolve the dropped call issue. Attached the txt, with the *60 occurring in the logs. Thanks in advance for your help Snom320 - Starcode 60.txt

I agree that it's hard to combat SPIT, especially with the design of toll free numbers. I won't be able to eliminate it but i can hope to try and minimize it. When i introduced the hangup time-out on the auto attendants it had a really positive effect for about a year up until now that they found a way to bypass the hang-up timeout. I think they use a rotating list of numbers and based on whatever metrics they gather they might single out numbers for repeat calling. Even if they keep calling back once every couple of days/months the time-out or uncondtional hang-up press will minimize damage to a couple of cents a month instead of 20-30$. It also clears up lines faster, if a call stays on the attendant for the maximum call duration of one hour then that line is inaccessible for that hour. Ultimately what i would like is an unconditional hang-up on an auto-attendant, button presses or not. Once the IVR message has played, timer runs for x amount of time and if no valid inputs have been used terminate the call.

Yeah the robots specifically target toll free numbers and sometimes they are attached to an auto-attendant as part of the call flow. I was able to modify the IVR so that the * isn't treat as a special key and passed it off into an extension with a disabled mailbox. But the main issue is that the same cannot be done for the # symbol and will actually just force the auto-attendant to keep replaying it's message. Also i wasn't able to replicate the redirection with my cell phone (tested with iPhone 5s) but it worked on a snom phone. So even if i get the * option to redirect with the fax tone detection i won't be able to do the same for # and i'm back to the same problem 1-2 months once the robots change up their strategy. I am however assuming how the robots stay active within the auto-attendants but so far using * and # was the only way that i found on how to beat the redirection and hangup options available to me. I know even if an uncondtional time-out is introduced into the auto-attendant it will not ultimately resolve the issue but it will minimize damage and make it a less lucrative avenue. When i introduced the hang-up timeouts it worked really well for a year and actually cut down on a lot of the spam calls. If you have any other suggestions i would be more than happy to read them. Thank you for your time.

Hello, In the past i noticed that robots would call on to auto attendants and stay there indefinitely or until the system drops the call. To circumvent that i started designing all the PBX's with a hang up time-out. This has worked well for a long time but i started noticing that the robots are able to stay on the auto-attendant for an hour before the system drops them. So i started doing my own little tests on how to bypass the hang-up time out. If you consistently press * or # the auto-attendant will keep your call connected without dropping you. Now i know that you can technically force them to drop by creating a F input for the * press but there is no similar option to the #. I can start implementing the F input in all auto-attendants but this might not work for attendants that are already using every single input option, and i would need two if to implement the #. I'm wondering if there's any setting or configuration that i can apply to the auto-attendant in order to beat these robots. It's not practical to blacklist as everyday it will be a new number. This is mostly present on auto-attendants that have a toll free number attached to them. Would it be possible to create some unconditional hang-up timeouts or maximum call duration on an auto-attendant? Thank you for your time. (Running version 5.2.5a)

Hello, A customer of ours has the Snom Meeting Point and they are experiencing one way audio on outgoing. It isn't a network related issue as there are over 60 sets on the same site without any one way audio. I've narrowed down the issue to the microphone but i think i'm missing something because it still actually works but not for speech. It's currently on the latest firmware 8.7.3.25. So basically the Meeting Point is unable to pick up speech from anywhere in the room, close or far away from the speaker. You are able to lightly tap around or on the microphone and this will actually be picked up by the Meeting Point. I received verification that this work as customer called himself and left a voicemail of him tapping on the microphone and listening to it afterwards. Is there any sort of setting that i am missing or is the microphone faulty? I've reset the device and was able to provision it. i've tried manually selecting the appropriate codecs but so far i'm only able to get the tapping noise to transmit.

Morning, I've just tested the paging and it seems to be working fine after a reboot. I'm not sure what the PBX does or doesn't do behind the scenes but i guess after a reboot it started doing whatever it needs to in order to alert the phones to pick up the paging call. I did compare the logs on the test calls i've made from both PBX's and it looked like on the bad server it wasn't allocating for call ports or opening sockets (during the page) but i don't know if this would be the cause of the issue or how to remedy it for the future. Thank you for your help.

Thanks for your response. How would i troubleshoot the phone not answering on the server end? I exported the exact same domain with the exact same equipment to a secondary server with the same server configuration and it works without any issues. There is no constant ringing and no timeouts required. My issue is that out of three servers only one experiences an issue with paging. i guess i'll have to reboot the server at night and hope it solves the issue. Will update the topic on Monday. Thanks.

Hello, I've run into a problem with one of my servers where the paging extension rings all the phones instead of performing a page. So scenario is like this: Ext 100 calls paging extension 300 > 101, 102, 103 Instead of the phones being paged they just ring as if it was a call to a hunt group, and if you answer it proceeds with the page. I've tried exporting the domain to two different servers to replicate the scenario and i seem to be experiencing the issue only on one server. My servers are running 4.5.1.1107 Zeta Perseids. Have you experienced an issue like this before? What do you recommend i do? Is there something i should do to avoid this in the future? Thanks for your help.

Ok follow up question. Before this i tested the BLF (Monitor, pickup, speed dial) and Speed Dial paramater button setting but the GXP1405 does not pick this configuration up. I have one setup as a BLF to an extension on the PBX and the speed dial to call a cell phone, but after the phone reboots the buttons still function as Line 1 and Line 2. Is there a specific paramater or value i have to use for the button number for the GXP1405 to pick up the configuration?

Hello, So i followed the wiki guidelines on how to provision the Grandstream Phones and was able to provision it without any issues. The GXP1405 doesnt' have a lot of button configuration (only two lines buttons), but i need them to work as park orbits. I tried testing several of the button features, BLF, speeddial, service flag..etc but the button settings from the PBX don't seem to affect the phone settings. Once i save the button profile it will cause the phone to reboot but other than that no changes are made. I thought to customize the template under the specific extension but the grandstream-gxp1400.xml is only present under the domain Customize section. Please let me know if there are any other steps i should take in order to configure the buttons remotely. Thank you

Hello, Ultimately what i want to do is painlessly transition my customers from version 3 PBX's to a version 5 PBX without having to touch the phones physically or to upgrade their respective servers. What i want to do is completely take down a version 3 server and change the DNS records and point them to a version 5 server. For the purposes of this discussion let's say 100.xxx is version 3 server and 200.xxx is version 5 server The FW of the phones on 100.xxx is 7.3.30 I will explain the procedures i take in my migration. 1. Export the domain from 100.xxx and import to 200.xxx 2. Obtain MAC of the phone and attach it to it's respective extension 3. Change DNS records to point testdomain from 100.xxx to 200.xxx 4. Reboot the phone once records change 5. Phones attempt to obtain configuration file but give return code 500 and ultimately fail 6. I upgrade the FW of the phones to 8.4.18 7. Reboot phone and it picks up configuration file and register on 200.xxx without issues What i'm looking for is guidance on how to do this is seemlessly without any issues. Preferably i would like to not have to push any updates to the phones on 100.xxx that has to upgrade the FW to 8.4.18 but so far it looks like this is the only way to transition the phones from 3 to 5. I would welcome any suggestions or direct me to the right path i would appreciate it, Thank you for your time.