sip

Hello everyone, I'd like to share an open-source project I've been working on for the Asterisk community. CloudSIP is a WebRTC SIP Phone that allows users to make and receive SIP calls directly from a web browser without installing a traditional softphone. It is available both as a...

you can delete thank you

Hello, We are experiencing a common, yet problematic, SIP dialog flow between our Freeswitch (v1.10.10) and the Zoom SBC, which frequently results in hanging calls. We are seeking insights into why Freeswitch fails to terminate the session properly on the first attempt. The Problematic Call...

I am a VoIP engineer working with open-source projects like FreeSWITCH, Kamailio/openSIPs, RTPEngine, some other testing tools like SIPp. My workflow includes: Kamailio cfg language for SIP routing logic FreeSWITCH XML configuration Lua scripting for advanced freeSWITCH dialplan apps Python/Go...

I have details of server ip, username and pass and I am able to login into zoiper but what I want to know is who is the voip/did provider? Is there a way to find it with these details as the server ip provided is magnus billing but I want to know the provider like if it's twilio, telnyx or any other

Hello. CustomPBX an open-source API server and web GUI for FreeSWITCH. It represent API for FreeSWITCH XML config structures stored in postgres database and generated for xml_curl module. Allowing to configure everything native FreeSWITCH can do but over GUI or API. It can import...

Hello guys, PBX: IncrediblePBX 2021.01 for Debian 10.9 Buster. Asterisk 18.2.1. Incredible GUI 15.0.22.33. Is it enough to make the RTP port range through the GUI, or do I need to run anything else in the server terminal? I've experienced some calls that when picked up, I can't hear anyone...

I am looking for voip/spoof that works to Belgium with a phone number that I choose if this is not allowed here my apologies for this

Hey sup everyone, new to the forum here. Just need some help here a bit, regarding something that had me banging my head for days now.. I got myself some credits on illyvoip and I'm able to make calls through X-Lite softphone client. Everything seems to be working well, I'm able to make US...

Hi everyone, We are currently facing a critical issue in our network of GSM gateways and SIM banks dedicated to VoIP GSM termination. Our SIM cards are unexpectedly ceasing to make calls, and we're seeking the community's expertise to help us troubleshoot and resolve this problem. Here's a...

Hello to all, I'm going crazy to use the t38modem (and HylaFAX) with Asterisk: the modem is able to answer the (FAX) call, but it doesn't accept the reINTIVE to switch codec to T38 and it sends 488 Not Acceptable Here to Asterisk. I installed the T38modem simply using the repository and the...

Hi to all, I have 2 Cisco phones, 7911 and 7975G and I'd like to use them in my asterisk pbx along with an old Cisco pbx At first I'd like to know if both phones are able to have an add-on SIP account and then how to do it. Thanks.

Specs: CentOS 7, Incredible PBX 2020 Machine, Fresh Install. IPv6 is disabled within CentOS. This is a public facing pbx and the public script has been ran. Some sip clients are only being detected by their ipv6 address and not the ipv4 and might not play audio. Everything works fine over...

Hi, Is there a standardized way to specify from a SIP client to a SIP server to temporary or permanently block incoming calls for specific numbers/addresses ? Basically block incoming invites.. Ideally by giving various options on how to block them (like not found, busy etc). SIP has tons of...

I have a yealink t21 how do I apply a free moh before it says [email protected] minus my sip account nothing plays help pkesdr

Hi There To Connect Me, a service from OKay, offers unlimited minutes to USA48 (not Alaska, not Hawaii) and Lower Canada (not Northern Territories) within the number of channels hired. For those who may be interested in some numbers: 14.99 USD the channel monthly, divided by 30 days = 0.4996...

Hello, I've read several on-line posts and forums on how to change the default ring time for SIP extensions. I've tried changing "Settings - Advanced settings - Ringtime default value to 5, but this didn't change the ringtime. I have also tried changing the " Extensions - my extension -...

Hello everybody.. I have elastix server and nortel ip phone 1120e sip phone , I have created domain in ip phone :.. Server.. 10.0.0.43 which is elastix server and it taked ip from dhcp server.. I made a ping from sever to phone and it replied... Also ping from phone to server and gateway and...

Hello I install chan dongle (Asterisk 13.23.1, Huweai e1550, firmware 11.609.20.03.356). Codec uLaw, g722, gsm I run traffic capture at Wireshark and it shows RTP send receive both way. But other caller dose not hear anything.

Hi! We're the developers of a new VoIP app, Sipnetic. If you use an Android SIP client, such as CSipSimple, Zoiper, or Linphone, you should check out Sipnetic. I'll be glad to provide any info about this app or SIP protocol in general. Google Play...