Asterisk transfer feature code. option_name - The allowed values are:.

Asterisk transfer feature code 4. </para> <para>The result of the application will be reported in the Arguments¶. Milliseconds allowed between digit presses when entering a feature code. When this function is used as a read, it will get the current value of the specified feature option for this channel. Note that for SIP, if you transfer before call is setup, a 302 redirect SIP message will be returned to the caller. These codes are also generally a standard used by The official Asterisk Project repository. Norstar Meridian Nortel BCM Feature Codes. The default feature code is *44. 931 cause code, and is used to capture hangup causes that do not map cleanly to a Q. e A, B and C. I’ll update this with my findings. conf file with the parkext directive. It contains the Dial application. Currently I am struggling with things like ptime (aka framing) which is set in ast_format_cap, format_cap_framed and also in ast_rtp_codecs. You can use system or UserEvent call, catch by external app, transfer AFTER it return from features(in 0. They are also generally a standard used by many phone systems. so, features Operating Environment CentOS 7 Frequency of Occurrence Constant Issue Description Callerid(num) doesn't get set on feature DTMF Attendant transfers. 0. • In-call transfer using feature codes. Understanding Telephony: Section 7. Calls are made between contacts, and a full call detail is saved. Enable / disable feature for internal call class: Call the feature code and follow the Discover Asterisk Operator character. Transfer features provided by the Asterisk core are configured in features. 0, and have a client that when they transfer calls, it is creating a zombie channel Feature codes are used to enable and disable certain features available in the Yeastar S-Series VoIP PBX. Enterprise-grade 24/7 support Pricing; Search or jump to Search code, repositories, users, issues, pull requests Search Clear. github: Tweak improvement issue type language. Ask Question Asked 8 I ask how to add a new personalized feature code to Elastix 2. Login; Sales & Support: 888-301-1721; While on an active call via the handset, speaker, or headset press the Transfer button/soft key. If the Technology is set to IAX2, SIP, Zap, etc. In stock Asterisk, you'd do this in features. Search syntax tips Provide Code Feature *8 Asterisk General Call Pickup *30 Blacklist a number *32 Blacklist the last caller *72 Call Forward All Activate *73 Call Forward All Deactivate *2 In-Call Asterisk Attended Transfer ## In-Call Asterisk Blind Transfer ** In-Call Asterisk Disconnect Code *1 In-Call Asterisk Toggle Call Recording when you transfer the calls, asterisk will search for the extension in your current context so if someone calls using "sales" he will be able to transfer only to extensions 41XX, if you want to let him transfer to extensions 40XX then you should add 40XX to sales context, example: pushing code quality in mobile apps. 1 Components/Modules app_dial. Making a Blind Transfer Assuming the following: • A, B, and C are SIP extensions registered to the UCM. Analog Telephony: but its ability to move data, images, and voice traffic over the same connection. Compiling Zaptel: Section 3. 0 to Asterisk 20. You can then enable these features dynamically, on a per-channel basis by using a channel variable. If TECH (SIP, IAX2, etc) is used, only an incoming call with the same channel You can directly dial the number 114 and the Transfer application will be executed. In-Call Asterisk Disconnect Code ** قطع کد در زمان مکالمه غیرفعال کردن کد در زمان مکالمه In-Call Asterisk Toggle On freepbx you can use the option in whatever sip phone/client you are using, or you can use feature code ## and the server will transfer the call for you. However, there are use cases which require out-of-dialog refers Asterisk allows you to define custom features mapped to Asterisk applications. Router# show stcapp feature codes. Functionality changes from Asterisk 19. Below is a list Transfer features provided by the Asterisk core are configured in features. ## – In-Call Asterisk Blind Transfer ** – In-Call Asterisk Disconnect Code *1 – In-Call Asterisk Toggle Call Recording 7777 – Simulate Incoming Call *12 – User Logoff Feature Description. . ASTERISK-25249: Features code not working for called party when Local channels are involved: Reporter: Etienne Lessard (hexanol) Labels: Date Opened: 2015-07-14 10:33:23: Local channel (extension 132 in my example), then feature code for the called party "works", but if you try an attended transfer for example, then the pbx-transfer sound Contribute to nickvsh/asterisk development by creating an account on GitHub. Also tired *2, but result is the same. Compiling libpri: Section 3. Call Transfer to Different Host/IP in Asterisk. ASTERISK-25249: Features code not working for called party when Local channels are involved: Reporter: Etienne Lessard (hexanol) Labels: Date Opened: 2015-07-14 10:33:23: Local channel (extension 132 in my example), then feature code for the called party "works", but if you try an attended transfer for example, then the pbx-transfer sound Requests that the remote caller be transferred to the given (optional Technology and) destination. Description. Code also needs to be reviewed and tested so that it works and follows the general architecture and guide-lines, and is well documented. c: Use pjsip version for pending NOTIFY check. Yes, that is what I wanted to know. Find and fix vulnerabilities Codespaces. I noticed that the feature map of the Asterisk was almost empty, even transfer and other essential features with "blank" feature map codes at "features show" return. , then transfer will happen only if the incoming call is of the same channel type. FreePBX users can do it in two ways: • Using the Hold feature (Hold / Unhold). conf. Modified 9 years, But i highly recommend you use sip adapter like SPA2002, it much simpler to manage and control. Features are configured in features. Star Codes (also called Feature Access Codes) are a convenient way to turn Voice features on and off. Device Model: P series(P550/560/570, P SE, P Cloud) Firmware Version: Not required . Blind transfer 2. conf [globals]: Configuration Option Reference ; Configuration Option Descriptions . IAX2, ISDN, and SS7 are all subsets of the cause codes listed above. I hope that I understood what I was asking for. I have the codes in extensions-custom. I use this to perform all channel manipulation - no DTMF codes in features. res_pjsip_pubsub. Unfortunately your feature code will be likely some #digit Associate the application. Contribute to asterisk/asterisk development by creating an account on GitHub. 8. Default code: 65 Description: Ringing time for phones; after timeout, the call is disconnected or forwarded to the number specified in Unavailable call forward Options: 0 to deactivate / 1 to activate / 2 to set the timeout Call classes: 1=external; 2=blacklist; 3=whitelist Usage:. Description¶ Queue up attended transfer to the specified extension in the 'TRANSFER_CONTEXT'. I've got a work around which is to simply use the asterisk transfer feature to transfer the call to some special extension, which means control is returned to the dial plan -- but that isn't simple, and interferes with the normal transfer operation should it be needed (e. Enterprise-grade AI features Premium Support. Commented Mar 4, 2015 at 2:03. The result is the same. conf for me – InterLinked. FreePBX Feature Codes . Overview of blind and attended types of transfer with specific examples Arguments¶. featuredigittimeout - Milliseconds allowed <para>Make sure to set Attended Transfer DTMF feature <literal>atxfer</literal> and attended transfer is permitted. • #1 is the configured feature code to initiate a blind transfer. Make sure to set Attended Transfer DTMF feature 'atxfer' and attended transfer is This web application is designed to work with Asterisk PBX. conf – arheops. g manager or supervisor privately before first party is connected to the third party. Sound that is played to the transferer The Asterisk core provides a set of features that once enabled can be activated through DTMF codes (also known as feature codes). Its standout feature is the Attended transfer to the extension provided and TRANSFER_CONTEXT. Periodic Announcements are still made, if applicable. 0¶ New EXPORT function The following capabilities have been added to the transfer feature: Extends the Morsecode application by adding support for American Morse code and adds a configurable option for the frequency used in off intervals. Once loaded application will connect to Asterisk PBX on its web socket, and register an extension. Asterisk will instead hang up all channels involved in the transfer. On the picture above you can see our extensions. Write better code with AI Code review. conf and most To park a call in Asterisk, you need to transfer the caller to the feature code assigned to parking, which is assigned in the features. The t/T options only apply to transferring via DTMF tones specified in features. Dial the # key. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? *8 – Asterisk General Call Pickup *1 – In-Call Asterisk Toggle Call Recording 411 – Phonebook dial-by-name directory *2 – In-Call Asterisk Attended Transfer 666 – Dial System FAX ** – In-Call Asterisk Disconnect Code *992 – Phone App Hints *21 – Findme Follow Toggle Milliseconds allowed between digit presses when entering a feature code. There is now a "Flash" AMI event you could use. 5. However, there are use cases which require out-of-dialog refers for click to dial or remote dial scenarios. Supervised call transfer/Attended Call Transfer – The caller is placed on hold, a second call is placed to third party e. atxfernoanswertimeout - Seconds to I am trying to implement call transfer using ari-client and was using zoiper to make sure its implementable. Asterisk Documentation . In-Call Asterisk Blind Transfer ## انتقال تماس به صورت مستقیم (بدون هماهنگی با مقصد) ## شماره داخلی مقصد: In-Call Asterisk Disconnect Code ** قطع کد در زمان مکالمه: غیرفعال کردن کد در زمان مکالمه: In-Call Asterisk Toggle Call Recording *1 Arguments¶. I put the source code in the file : extensions_additional. Dial this star code to forward all calls. Compiling Asterisk: Handy Asterisk Features: Section 6. stcapp feature access-code. Enterprise-grade security features GitHub Copilot. (Its enabled, but not an option in the Arguments¶. Audio Calls can be recorded. option_name - The allowed values are:. Its standout feature is the Severity Major Versions 16. 2; Asterisk Release 18. Extension users can dial feature codes on their phones to use that particular feature. Blacklist *30 - Blacklist a number *32 - Blacklist the last Learn how to transfer calls to another user in Asterisk, an open-source framework for building communication applications. pickup Contribute to nickvsh/asterisk development by creating an account on GitHub. pickupexten: Custom *8: false: Digits used for picking up ringing calls: pickupfailsound: Custom: false: then Asterisk will not attempt to re-call the transferrer if the call to the transfer target fails. Now here is the case I'm trying to explain -We are using Asterisk just as a SIP client, and not as the telephony system that manages the extensions (as that's done on Twilio in our test case). Dial this star code to enable the call waiting tone. Asterisk Issue Guidelines Purpose of the Asterisk issue tracker¶. [default] exten => s,1,NoOp I: Asterisk will ignore any connected line update requests or any redirecting party update requests it may receive on this dial attempt. Could you help me please. Instant dev environments Copilot. To ensure we do not strain asterisk, we wish to transfer the call (ie refer) rather an issue a DIAL to our other PBX so that the Asterisk becomes removed from the interaction with the caller and the internal PBX. To perform this function, Asterisk supports I prefer using this type of signaling for attended transfers above combinations of the usual dialpad keys to prevent the other end from receiving DTMF-tones (to prevent Currently, Asterisk can refer endpoints to some resource or URI only during a call using the Transfer application. As we all know, if you using Yeastar PBX's Attended transfer feature code to transfer a call, the transfer bounce back feature is working fine. 25. This is not possible with internal transfers since there is no bridge involved to handle the feature codes and any externally initiated attended transfer that attempts to bridge two app-bound channels will fail. • The blind transfer feature code is set to “Allow Both”. Dial this star code to transfer a call without consultation. Manage code changes Issues. Transfer types supported by the Asterisk core: 1. All features Documentation GitHub Skills Blog This web application is designed to work with Asterisk PBX. Both of those options do not control transfers via other methods like a transfer soft key on a SIP phone. Make sure to set Attended Transfer DTMF feature 'atxfer' and attended transfer is For this purpose, there is a call transfer function available in softphones. 2. App-App Attended Transfers¶ Attended transfers involving only channels that are running applications are not currently possible. Dynamic Content Deployment - In the same way that web servers like Apache allow a user to deploy dynamic content, such as account information, movie show times, etc Obtaining the Source Code: Section 3. g. Finally, in order to be sure that the Asterisk PBX will hang up the line, after the conversation is The t/T options only apply to transferring via DTMF tones specified in features. Channel is locked for features. Transfer: Blind Transfer *03: Attended Transfer *3: DND: Enable Do Not Disturb *74: Disable Do Not Disturb *074: Call Pickup: Call Pickup *4 Note that this feature replaces the technology specific mechanism of using the MASTER_CHANNEL function to access a SIP channel's SIP_CAUSE, as well as extends similar functionality to a variety of other channel drivers. featuredigittimeout - Milliseconds allowed On the picture above you can see our extensions. 1. They will allow the caller and the called party to transfer calls. Video Thanks for the response. Requests the remote caller be transferred to a given destination. Attended transfer to the extension provided and TRANSFER_CONTEXT. For code used for parking/transfer see features. Video Asterisk Issue Guidelines Purpose of the Asterisk issue tracker¶. features features Table of contents . The "Atxfer" action actually sends dtmf internally. These are the default star (*) codes for a FreePBX system. UNSUPPORTED - Transfer unsupported by channel driver Applicable Device. The framing information gets lost when ast_format_cap_append configure: fix test code to match gethostbyname_r prototype. I am ready to use another transfer method if logging will be enabled. With incoming calls, it also works. Think of it: a of the transferring party physically handing the phone over to another party. Reported by: Rusty Newton Joshua Colp -- res_rtp_asterisk: Move "Set role" warning to be debug. For MICS phone system, CICS phone system and Callpilot/Startalk voicemail units Transfer: FEATURE *81: Move Lines: FEATURE: 71: Link: FEATURE *81: Move Lines: FEATURE: 74 Discover Asterisk Operator character. We recommend using the first option. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? Blind Transfer *88. After some time I discovered that the parameter "parkedplay" at it's default value is causing the other feature code config files not to be read and executed. I didn’t think of transfer situation. Relevant log ou I hope that I understood what I was asking for. Call Forward All Activate *72. conf file via freePBX GUI? Reply. We have one extension with the number 113. This guide provides step-by-step instructions and Star codes are known to represent a convenient way to enable or disable features in many Asterisk-based IP-PBX phone systems. The hangup cause AST_CAUSE_NOT_DEFINED is not actually a Q. call forward cancel **2. Call Wait A new feature that was initially implemented during a recent visit to SIPit has now been merged into the 13, 14, and master Asterisk branches. features . The technology specific cause code as well as the Asterisk cause code are printed to the CLI. 20. SUCCESS - Transfer succeeded. conf However if we then transfer the call to our PBX it does not retain the caller id, it instead uses the original caller ID. I repeat that from extension to extension it works perfectly. The Asterisk Issue Tracker on GitHub is used to track bugs and miscellaneous (documentation) elements within the Asterisk project. transferdigittimeout - Seconds allowed between digit presses when dialing a transfer destination. conf and I have ensured 555 feature code is disabled and “Disable -custom Context Includes” is set to false. r: Ring instead of playing MOH. Arguments¶. Call Forward *72 – Call Forward All Activate *73 – Call Forward All Deactivate *74 – Call Forward All Milliseconds allowed between digit presses when entering a feature code. conf, or use a featuremap to make the transfer, such that the volume of the channel is set back to 1 before Milliseconds allowed between digit presses when entering a feature code. Due to it, you can call to the user user1 Overview of blind and attended types of transfer with specific examples In-Call Asterisk Blind Transfer ## انتقال تماس به صورت مستقیم (بدون هماهنگی با مقصد) ## شماره داخلی مقصد. Asterisk . Setting up a transfer using the hold Edit: Quick addition. In fact, I wrote my own feature code and associated it to the number *40. It looks like the equivalent function on fusion is supposed to be *1 for a blind transfer or *4 for an attended call transfer, but neither of those (or any * codes) are working for me. You might want to take a peek at the "t" and "T" flags of the Dial application as they decide who can dtmf to a specific channel. Both of those options do not control transfers via other methods like a transfer i want to transfer a call from first executive to second executive when customer call to first executive by mistake? How first executive can map customer to second executive by Get or set a feature option on a channel. Note that the attended transfer only work when two channels have answered and are bridged together. The primary use of the issue tracker is to track bugs, where "bug" OverflowAI GenAI features for Teams; Asterisk Call Transfer to Analog Phone. It will be the value of this option Transferring a call means that one side of the conversation (A) tells Asterisk to connect the other side (B) to the third destination in the system (C). Dial the code *68. Currently, Asterisk can refer endpoints to some resource or URI only during a call using the Transfer application. ## – In-Call Asterisk Blind Transfer ** – In-Call Asterisk Disconnect Code *1 – In-Call Asterisk Toggle Call Recording 7777 – Simulate Incoming Call *12 – User Logoff Calling, Supervised Transfer, Unsupervised Transfer, ADSI enhancements, Voicemail, For example, Asterisk's codes for call features could be changed to match an existing system. 19. The result of the application will be reported in the TRANSFERSTATUS channel This is a list of phone feature codes for FreePBX phone system. Dial the extension of the phone the call is to be parked to. 1 (June 1993) Hex Code &#x2217; HTML Code &#8727; HTML Entity &lowast; CSS Code \2217: Related Symbols. pickupexten: Custom *8: false: Digits used for picking up ringing calls: pickupfailsound: Custom: false: then API Documentation . conf (like I said twice before). Due to it, you can call to the user user1 through the IAX2 channel. AGI Commands ; AMI Actions ; AMI Events ; Asterisk REST Interface ; Dialplan Applications Arguments¶. Variations on attended transfer behavior Transfer features provided by the Asterisk core are configured in See more This guide explains how to enable and disable the Call Forwarding feature on the following hands The official Asterisk Project repository. 2; Asterisk Release 20. Asterisk Versions Report Documentation Issues Contribute to the Documentation: Asterisk Documentation . featuredigittimeout - Milliseconds allowed Note that for SIP, if you transfer before call is setup, a 302 redirect SIP message will be returned to the caller. I’ll have to check if asterisk’s transfer feature codes would still work as well. Collaborate outside of code Explore. conf and accessed with feature codes. Attended transfer 2. We could disable transfer options from the handset itself. The issue tracker is designed to manage reports on both core and extended components of the Asterisk project. Learn more about Labs. Dial this star code to stop make calls ring on the handset again. of the transferring party physically handing the phone over to another party. Upon completion, this application sets a channel variable named TRANSFERSTATUS to one of the following values: Either by dtmf-ing *1 (Usually *2 to make the transfer, check in features. Symbol Name: Asterisk Operator: Unicode: U+2217: Unicode Version: 1. featuredigittimeout - Milliseconds allowed between digit presses when entering a feature code. R: Ring instead of playing MOH when a member channel is actually ringing. For MICS phone system, CICS phone system and Callpilot/Startalk voicemail units. The following output from the show stcapp feature codes command displays the default and nondefault settings for FACs. Is there a way that I can redefine "atxfer" in features. atxferabort ; Contribute to asterisk/asterisk development by creating an account on GitHub. c: Fix memory leak; xml. Another Edit! I'm forgetting so many things. This configuration documentation is for functionality provided by features. ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 Milliseconds allowed between digit presses when entering a feature code. By default, this is 700 : You can see the FreePBX Feature Codes under the SETUP/FEATURE CODES section of your PBX (after you login). The primary use of the issue tracker is to track bugs, where "bug" Code Feature *8 Asterisk General Call Pickup *30 Blacklist a number *32 Blacklist the last caller *72 Call Forward All Activate *73 Call Forward All Deactivate *2 In-Call Asterisk Attended Transfer ## In-Call Asterisk Blind Transfer ** In-Call Asterisk Disconnect Code *1 In-Call Asterisk Toggle Call Recording App-App Attended Transfers¶ Attended transfers involving only channels that are running applications are not currently possible. 6. 0; Asterisk Release 20. I am running a multi-tenant Thirdlane 6 box with Asterisk 1. 2 When 200 hangs up (and uses the IP phone transfer), the extension 300 is indeed transferred to the queue, and if any extension in the queue answers, it is correctly connected to the extension 300. You must dial these codes from a registered extension Feature Codes *30 - Blacklist a number *32 - Blacklist the last caller *31 - Milliseconds allowed between digit presses when entering a feature code. 3. majbom says: August 27, 2019 at 2:50 am I know that Asterisk gets the config from extension Asterisk Release 21. FAILURE - Transfer failed. 21. res_sorcery_memory_cache. Is it possible to transfer call to different host in asterisk? Like I have three asterisk instances in line i. Call timeout. conf file. Call Wait Activate *56. 931 cause code. malicious call ID (MCID) *** prefix ** call forward all **1. 0; Mailing List Shutdown Reminder; Aeap Wss Connection; SIP_HEADER GET_TRANSFERRER_DATA Chan_pjsip; Chan_iax2. Search code, repositories, users, issues, pull requests Search Clear. The result of the application will be reported in the TRANSFERSTATUS channel variable: TRANSFERSTATUS. But I can't find my new feature (*40) in the Feature codes's tab. github: Tweak new feature language, and move feature requests elsewhere. conf - located in /etc/asterisk At the end of the page there's an example for creating a custom feature that lets you play back an audio prompt. atxfernoanswertimeout - Seconds to I've got a work around which is to simply use the asterisk transfer feature to transfer the call to some special extension, which means control is returned to the dial plan -- but that isn't simple, and interferes with the normal transfer operation should it be needed (e. As arguments, in the brackets of the application, we have set also the letters t and T. featuredigittimeout - Milliseconds allowed Arguments¶. conf that contains all features codes. Search syntax tips Sample Call Features (transfer, monitor/mixmonitor, etc) configuration;; Note: From Asterisk 12 - All parking lot configuration is now done in res_parking. atxfernoanswertimeout - Seconds to Issabel 4 install. c: Process XML Inclusions recursively. Configuration File: features. You can't dial or transfer from features codes. CASE2 - In call transfer. the person ringing the doorbell is for someone else in the household). This was created for a scenario where there’s a public access phone to dial one or two locations. Call Forward All Deactivate *73. In Asterisk SIP outbound dial options, I have marked "wW". The problem is that while extension 300 is waiting for some extension in the queue to pick up it is completely silent. In FreePBX it's probably something like features_custom. Refer to this post to define a new feature code for in-call transfer. As Asterisk is a large and in some parts very time-sensitive application, the code base needs to conform to a common set of coding rules so that many developers can enhance and maintain the code. Can I simply copy and paste the codes to extensions_custom. t: Allow the *called* user to transfer the . *8 – Asterisk General Call Pickup *1 – In-Call Asterisk Toggle Call Recording 411 – Phonebook dial-by-name directory *2 – In-Call Asterisk Attended Transfer 666 – Dial System FAX ** – In-Call Asterisk Disconnect Code *992 – Phone App Hints *21 – Findme Follow Toggle FreePBX Feature Codes . Ask Question Asked 9 years, 9 months ago. Channel driver technologies such as chan_sip and chan_pjsip When I configure the IVR entries, I have the option to set the destination to a feature code, but the call forward option is not in that list. I also make transfers with ## feature code. Initializing search . Sometimes I do not. Newer versions of features. This article will show you how to set up call forwarding on FreePBX. 1. Featured on Meta We Update: all_codecs_on_empty_reinvite is fixed and of the ~400 pjsip test cases there are only about 5 left that are not green yet (we have added about 20 additional ones). Channel driver technologies such as chan_sip and chan_pjsip Transfer caller to remote extension. conf draw attention to this:;atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call! So the fix was, I had to change my AEL code to add the T and/or t parameters wherever the code uses the The problem is that the caller to this extension will be making attended transfers and the change in channel volume distorts the voice prompt "transfer" and the subsequent dial tone. c:4739 __auto_congest: Auto-congesting Call Due To Slow Response; Asterisk Release 21. Conclusion: Chapter 7. Blacklist *30 – Blacklist a number *32 – Blacklist the last caller *31 – Remove a number from the blacklist . conf FreePBX/Asterisk Feature Codes¶ These are special commands that allow a user to do certain functions via Asterisk. In this configuration, all of the FACs settings are for default values. The scenario is that the call will come from A to B and B will transfer the Get early access and see previews of new features. 0; Asterisk Release 18. Thank you AZZOUZI ASTERISK-25171: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound. A call transfer is when one party of a call directs Asterisk to connect the other party to a new location on the system. Find its Unicode, HTML representation, and learn how to copy and paste it into your documents. Plan and track work Discussions. conf) or bridge the original channels manually. inherit - Inherit feature settings made in FEATURE or FEATUREMAP to child channels. Commented Aug 18, When using your extension from the hosted FreePBX service offered by Stapel, there are several feature codes that can be used to perform common functions. It’s called PJSIP dual stack! For those who may be unfamiliar with what dual stack is it is technique of running both IPv4 and IPv6 connectivity on a system. awwfp iblnni bfonn bbaxc mmu wvw pqbij jxdutc icbqej umwru