I am using asterisk 15.5 as voip server and twillio trunk to make outgoing and incoming call but when I hangup on an incoming call to sip client then 603 Declined event is coming to asterisk but after 4-5 sec again I am getting incoming call repeatedly. is it the issue with twilio trunk or 603 delined does not getting propogated?
603 Declined event is coming while hangup on incoming call with sipML5 but after 4-5 sec again I am getting incoming call
1.9k Views Asked by sirvisuresh At
1
There are 1 best solutions below
Related Questions in TWILIO
- Twilio Salesforce integration (Chat Transcripts)
- How to set up "No Input Prompts" for whatsapp bot which is developed on Google Dialogflow CX and deployed through Twilio
- How to add context to chat-gpt twilio customer service bot?
- Stop Audio Stream on Twilio WS - Python
- Why does Twilio Studio insert line breaks around variables in my text and json?
- I want to auto update the Primary handler fails webhook Url using twilio functions, can someone guide me?
- Endpoint URL to connect the Dialogflow CX agent to WhatsApp using Twilio
- How to stream audio of <Dial> to websocket?
- Can Twilio save a voicemail in MP3 format when using the AWS S3 external storage option
- Cann't send back data using websocket for Twilio call
- Trying to send Twilio webhook to Kommo crm
- Twilio ASP.NET Web Api Request Validation - 'Specified method is not supported' When Attempting To Read From Content Stream
- Looking to trigger webhook on all api created sms sent messages w/o passing messaging channel or webhook parameter
- Migrating from whatsApp Template to Content Builder
- Twilio: is it possible to finish the <Dial> verb before timeout?
Related Questions in ASTERISK
- Call 2 numbers simultaneously from softphone (Asterisk - FreePBX)
- Need to connect my WebRTC stream(Handled by PeerJS) to my Asterisk server
- AsterNET.ARI how to implement a simple call between internal numbers?
- Make multiple calls asterisk ari python
- Amı Originate Calls Diffrent Context From Sended Context
- How to get pn-prid with PJSIP_HEADER in Asterisk dialplan
- Problem Getting RECOG_INSTANCE() Value When Using MRCPRecog with Asterisk-Java AGI
- custom live call monitoring function with nodejs/asterisk-manager not working as expected
- How do i write a dial plan that plays a audio file takes user input and plays another audio file?
- Using Cut with multiple delimeter
- asterisk dialplan, how to use includes in contexts right
- Cannot listen to an asterisk manager event : QueueMemberPause
- Asterisk AMI call originate with python for a group of contact at a time
- dialplan show the unicode char
- Prepending a 1 to outbound Caller ID FreePBX
Related Questions in BANDWIDTH
- Cause of Excessive SMB2 FIND Requests When Writing Files to a Windows Server?
- In the cpanel bandwidth report is it possible to change the colors of the graphs? the Red color
- Adaptive Bandwidth for MeanShift++
- Stream benchmark returns impossible bandwidth
- iperf 2.1.9 bidirectional test cant figure out the server-client flow
- How find the LTE bandwidth in Android
- Is bandwidth in networking different from bandwidth in webhosting?
- Is the network capacity calculation right?
- R Code for LU Factorization for a Banded Matrix
- What Happens When Font Awesome Bandwidth is used used more than the npm bandwidth allowed for my plan this month?
- How to test the “random access bandwidth" of memory?
- What is the problem in my code? (Bandwidth-Limitation)
- Reverse proxy reduces download speed
- Ways to reduce size of data transferred over a network?
- What is the problem with my code(bandwidth limitation)?
Related Questions in SIPML5
- SIPML5 - Problem changing audio input device
- sipml5 do not hang up automatically at the end of a call
- Call quality metrics in sipML5
- 603 Declined event is coming while hangup on incoming call with sipML5 but after 4-5 sec again I am getting incoming call
- SipML5 with Kamailio as Sip Server return 488 in make Call
- WebRTC + Adapter.js giving addremoteCandidate error while connecting audio call in MS Edge
- failed Connection closed before receiving a handshake response sipml5
- Video Conference MCU NAT Traversal not work
- No video in WebRTC call on smartphone(Android) via Asterisk
- How to configure REFER call in SIPML5 WebRTC?
- Call established but no audio on both ends in sipml5
- can I use indexDB to store sipml5 client objects
- No audio in sipML5 with Firefox 58
- Kamailio and sipML5 and unable to get a connection
- How can I get in SIPML5 the remote display name?
Trending Questions
- UIImageView Frame Doesn't Reflect Constraints
- Is it possible to use adb commands to click on a view by finding its ID?
- How to create a new web character symbol recognizable by html/javascript?
- Why isn't my CSS3 animation smooth in Google Chrome (but very smooth on other browsers)?
- Heap Gives Page Fault
- Connect ffmpeg to Visual Studio 2008
- Both Object- and ValueAnimator jumps when Duration is set above API LvL 24
- How to avoid default initialization of objects in std::vector?
- second argument of the command line arguments in a format other than char** argv or char* argv[]
- How to improve efficiency of algorithm which generates next lexicographic permutation?
- Navigating to the another actvity app getting crash in android
- How to read the particular message format in android and store in sqlite database?
- Resetting inventory status after order is cancelled
- Efficiently compute powers of X in SSE/AVX
- Insert into an external database using ajax and php : POST 500 (Internal Server Error)
Popular # Hahtags
Popular Questions
- How do I undo the most recent local commits in Git?
- How can I remove a specific item from an array in JavaScript?
- How do I delete a Git branch locally and remotely?
- Find all files containing a specific text (string) on Linux?
- How do I revert a Git repository to a previous commit?
- How do I create an HTML button that acts like a link?
- How do I check out a remote Git branch?
- How do I force "git pull" to overwrite local files?
- How do I list all files of a directory?
- How to check whether a string contains a substring in JavaScript?
- How do I redirect to another webpage?
- How can I iterate over rows in a Pandas DataFrame?
- How do I convert a String to an int in Java?
- Does Python have a string 'contains' substring method?
- How do I check if a string contains a specific word?
You can go under your Twilio Call Logs, click the particular Call(Sid) in question, and view the public packet capture (pcap), to see what Asterisk is sending back to Twilio. Twilio should not be resending the INVITE / advancing to the next origination URI (if configured) if receiving a 603 response from Asterisk, so most likely Asterisk is sending some other response code. You could also look at Asterisk logs to determine same.
Source: https://www.twilio.com/docs/sip-trunking#multiple-orig-uris "Note: If any of the following SIP status codes are returned ("2xx", "400", "404", "405", "410", "416", "482", "484", "486", "6xx"), Twilio will not fail over to the next origination SIP URI. If there is no SIP response from a given server, Twilio will fail over after 4 seconds."
It is possible the carrier is retrying. You should be able to see this if there are multiple CallSID's which indicates the carrier sent it to Twilio multiple times.