Q: How to make calls?
You can use either our web phone and make calls right from the browser or a third party soft phone application using web phone is very simple. Just press the phone icon near the user. You can also open the operator dashboard for a convenient interface for using the web phon. You can use either the web phone window or the web phone plugin which is integrated in the page itself. If you want to use a soft phone, you can download pre configured applications by pressing options for a user.
You can download Zoiper, PhonerLite and Linphone. If you would like to use another soft phone, you can find all the required data for registration by pressing on the SiP device number.
Please note that when using web phone, the SRTP mode must be on. If you're using a soft phone, you first need to disable SRTP. In case your soft phone supports SRTP, you can enable it in both softphone and teltel.
Q: How to receive calls in TelTel?
To receive calls, you will need to buy numbers. Go to the phone numbers tab and click on buy a number. Here you will find all available numbers. Please note that some destinations will require verification. If you're not sure what number to pick, contact our customer support. When number is added into Your account proceed to call-flow section to create inbound call-flow logic for Your number. Note that inbound calls will not work properly without incoming call-flow logic. To see how to build inbound call-flow, please address call-flow section of this manual.
Q: How to transfer call to other user? [*2]
Start Call Transfer: While on the call, press * and 2 on your phone's dial-pad consecutively. This tells the Asterisk system that you want to transfer the call.
Wait for Confirmation: You'll hear an audio message from Asterisk saying "transfer". This means the system is ready to receive the extension or phone number you want to transfer the call to.
Enter the Destination Number: Quickly enter the extension number or the phone number you want to transfer the call to. Remember, enter the numbers rapidly (within about 1 second between each digit).
Wait for Connection: Once you've entered the number correctly, the system will start calling the destination. The person you're transferring will be on hold. The person on hold will hear hold music.
Complete the Transfer: You complete the transfer by hang up once the person to be connected with picks up the phone. The person you had initial call will be now connected with the destination.
Q: How to create Predictive dialer campaign
Subscription Requirement: Ensure you're subscribed to the TelTel standard plan to create a Predictive Dialer Campaign. The standard plan allows uploading more than 5 contacts into the dialer.
Creating the Campaign:
Access the web interface and go to Tools/Autodialer. Create a campaign record and click "Edit" to set it up. Specify the campaign name, choose "Predictive dialer" as the type, and select caller IDs and users for the campaign. The campaign can use multiple caller IDs, and there's a feature for randomizing caller IDs based on settings.
User Groups and Functionality:
Utilize user groups for predictive dialing. The Predictive Dialer works alongside the operator infopanel, ensuring agents receive call information. Use the "Call only to Active operators" option to send calls to logged-in users in both the infopanel and phone application.
Call Handling and Settings:
Campaign Management: Manage campaign identification, known contacts, type, and caller ID settings. Permission Groups: Assign permissions and access levels for users involved in the campaign. Call Handling Settings: Define call limitations, call only to active operators, increase call count, and retry settings.
Customized Call Statuses:
Assign specific categories (General, Technical, Legal, etc.) to categorize call outcomes.
Configure autodialer settings and call distribution to operators. Define criteria for retrying calls based on various scenarios. Set post-processing pauses and manage features related to dial tones, call stopping, and handling answering machines.
After the record has been created and saved, click to upload contact and copy/paste contacts into the text field or upload contacts via CSV file.
This tool offers a comprehensive suite of features for managing outbound calling campaigns, allowing customization of settings for effective contact management, call distribution, and handling various call scenarios.
Q: How to create Robo-calls campaign
Robo-calls basically are set up in the same way as Predictive dialer, however instead of users, calls will be routed into the inward dialing number selected as callerID for the campaign. For that you will need a valid DID number. The sequence of events logic that will occur upon call reaching the client is set in the inbound call-flow logic builder.
Q: How to use dynamic caller id. [Caller id from my PBX or softphone]
Dynamic callerID allows client to set the callerID on clients end instead of TelTel web interface. Usage of Dynamic callerID must be first agreed with from the TelTel support end. By default this feature is disabled. When You have confirmation that dynamic caller ID is enabled on Your account, for the SIP device employed for such calls, You should set SIP ID of TelTel admin account as callerID. Now Your callerID's will be passed over our system. To set the caller id adjust your invite header with this FROM field: From: "<callerID>" <sip:sipid@server> or set it in callerID field in your softphone application.
Q: How long are my call recordings stored in your system?
By default call recordings are stored for 90 consecutive days. This setting can be amended on demand. Increasing the storage for a year will cost additional 1,5 Eur to Your overall price per user. You can decrease this value without charge by making inquiry in support or turn off the call recording in Your account settings. Call recordings can be purged by checking the “delete all call recordings" in Your TelTel account settings. Checkbox does not have immediate effect, but will be scheduled for midnight, based on server time, if checkbox is still checked at that point.
Q: Why are my calls to the same destination number rated differently? Can I fix my rates?
There may be several reasons for that. TelTel by default is utilizing dynamic rate system, which means that depending on provider the call takes rate may differ. TelTel always chooses the best possible provider as highest priority in terms of price and quality. In case if primary provider is unable to terminate the call, attempt is given to secondary, tertiary and so on provider en-route. This is happening within one single call attempt from your end and is made so to ensure highest possible answer rate.
Another possible reason is chosen callerID for the call. For instance calls from EEA callerID to EEA destination will be rated cheaper then f.x. call from Non-EEA to EEA which is deemed international while EEA to EEA - local.
Upon specific agreement with teltel, it is possible to fix your rates to a certain value, however, fixating on cheaper rates will exclude international routes and calls to specific mobile providers resulting into lower reachability.
Another solution is to set the desired rate ceiling in your teltel web interface, settings section. Thus calls above Your specified ceiling will be blocked and you will not end up having unnecessary high rate calls.
Q: I am unable to register my softphone, registration fails, what could be the reason?
Q: I am having one way audio issue, sometimes call gets disconnected after several seconds, sometimes I have dropped the call, however it continues on the other end. What is happening?
Very likely that SIP ALG is enabled in your local network router. Check SIP ALG in Your router settings. Google how to disable it on your specific router model in case if You don't know how to do that or contact out customer support for assiatance. In case if Your SIP ALG is disabled, router has been restarted and SIP ALG disabled state verified however issue persists, kindly contact our support with the call example for further assiatance.
Q: What is SIP ALG and why should I disable it?
SIP ALG, or Session Initiation Protocol Application Layer Gateway, is a feature found in many commercial routers. It helps users initiate SIP calls more reliably, even behind secure firewalls. However, ALG's network address translation (NAT) tool, while aiming to improve connections by changing private IP addresses and ports into public ones, can cause issues. Its erratic data packet modifications often disrupt modern VoIP calls instead of enhancing clarity.
Q: Is this system secure? How can I be sure that my data will not end up with the third party or will be lost?
We value our clients security. Your data integrity and overall security is our top priority.
TelTel is operationg under GDPR rules, therefore your data can be disclosed only to parties directly involved into your specifically requested TelTel service provision like database or service server hosting services or other voice or SMS providers we are connected to. For more information on this address your DPA addendum to your standard TelTel agreement.
To prevent possible losses of data TelTel has database system where several databases acts as one being constantly replicated. This means that if one database fails system continues to run on several other separate independent databases ensuring your data integrity and service continuity.
All our employees have several access levels based on their competences and are under NDA (Non Disclosure Agreement) ensuring that Your data is secured from internal leaks as well.
For the calls we use SRTP encription, so your calls cannot be sniffed within the network if SRTP is enabled. In case if SRTP is not an option, we have a VPN integration solution.
Q: How to set permissions and access for each group and user [Create group, set permissions, assign to users]?
Each user in TelTel system have their own credentials for teltel web interface access. By default only admin has access to section in teltel web interface, for regular users it will show no option but to log-out. in order to provide other users with access You must create permission group or groups. This is done from users section, permission groups.
When creating a group, pay close attention to hierarchy levels, they indicate whether this user has administrative rights above other users. In hierarchy levels lower number means higher hierarchy. This way users on level 10 will be above users on number 20. Users can be part of several groups, thus ensuring that Team Leaders can have their own specific groups assigned. In such case Team Leader group must have higher hierarchy then his team members, thus allowing TL to access his team data. Users on the same hierarchy level cannot see each other data even if they are part of the same group. Rest is quite self explanatory, in the group settings check the necessary permissions You would like to provide to the group and save. You assign group to users in users section, group column in users list. You must create at least one group only then option to assign group to users will appear in users section.
Q: I am unable to reach destination from TelTel, however from my mobile phone destination is reachable and picks up the phone. Why?
Routing issues are not uncommon in telco industry especially if you start to work with calls on industrial levels. There are many reasons for this to happen. In order to find out the exact reason, it must be investigated. For this you must contact the support with the precise call example. Remember, each call is unique, therefore precise example is crucial for swift and correct resolution. Precise example is a copy/paste of call details from Your call list containing date, time, callerID and dialed number. Or, you can report a call directly from the call list by expanding it by clicking on the arrow in the leftmost column of the call and clicking on “Report an issue with call”. Don't forget to add an issue description and we will check it ASAP.
Q: My call quality is bad, conversation is laggy, sometimes conversation partner sounds like a robot. How to resolve this?
This might be due to package lost. Check ping to the PBX server by writing a console command: “ping <domain>”. In case if latency is high there is a high chance that packets will drop and audio issue will be evident. We have several servers in different regions. In case of high latency we can switch You to other server. However it is a good practice to check Your internet connection overall, sometimes the cause of the issue is the specific internet service provider. You can check conduct your latency test here:
or web based:
In case if You are experiencing package loss above 1% on these tests, there is a high chance that you will hear this during your VoiP conversations. In this case please consult your ISP for further assistance. Otherwise If Your package loss is 0% or close to 0%, inform our support to assist with changing your primary server, other internet route might alleviate the issue.
Q: How can I connect my PBX or Asterisk to TelTel?
If you would like to connect your instance to our system, check TelTel voice and SMS API documentation in web interface settings, API and integration settings.
In case of “peer" connection, make sure that FROM field of SIP header you do send SIP ID of teltel user, as well, Your IP must be whitelisted on our end. Fro your convenience we suggest you to utilize the “friend" type of connection.
TelTel uses chansip, we are planning migration to pjsip in the near future.