sip

  1. P

    Persistent Issues with SIM Card Blocks

    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...
  2. M

    Anyone use t38modem with Asterisk?

    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...
  3. F

    Cisco phones reuse

    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.
  4. jbrandon.cj

    Asterisk detects ipv6 and doesn't play audio

    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...
  5. vasilakisfil

    Standardized way to block SIP Invites

    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...
  6. A

    TIPS Need help with moh yeslink music on hold please music@iptel doesn’t work

    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
  7. D

    DEAL Unlimited USA48 and Lower Canada Termination Trunk

    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...
  8. B

    Ring Time for SIP Extensions

    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 -...
  9. A

    QUESTION Ip phone problem with Elastix

    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...
  10. shetu

    TIPS One way audio

    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.
  11. ankyr

    FOOD FOR THOUGHT New SIP client for Android

    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...
  12. W

    TIPS Securing Server and Removing knockd?

    I consider myself somewhat technologically abled but working with Incredible PBX, VPSs, and networking definitely is getting my hands dirtier than I thought it would. I have several endpoints that have dynamic IPs and it's not really feasible to have them connect through a VPN. (1) I am...
  13. S

    SOLVED Incredible PBX 13 Extension will not register

    Hi All, I just built a new Raspberry Pi 2 with Incredible PBX 13.0.192.19 to replace an older version (PBX 11-3?). I did all of this on a second Pi so I could still have the old one up and running for reference. I gave the old Pi a new static IP address and the new Pi the static IP address of...
  14. wardmundy

    ALERT GV: The Sky Has Fallen... Really

    We've been down this Sky Is Falling Road several times before with Google Voice, but this time it looks to be real. Google has announced they are dropping XMPP support in mid-June. That's 5 weeks. All of the usual indicators that they may change their mind appear to be missing this time around...
  15. big charlie

    TIPS DID provider w/ SMS for Mexico City

    I need a DID for Mexico City that also has SMS capability for the same number. Vitelity does not offer SMS for international DIDs. And other services had could not offer both on the same phone number. Anyone have any leads. If you are not 100% sure, go ahead and give me the name and I'll check...
  16. J

    TIPS How to refresh registration manually in 3CX PBX edition (Free)

    I recently had to make some changes to my internal DNS, which caused a small outage. During this window, it appears it knocked out the registration on the server. Now I can't seem to get the SIP registration to re-register. I've rebooted the server to try and force it, but it seems it just...
  17. S

    SOLVED Problems with both out and inbound calls

    Hi. Installed XIVO a few months ago, got it working with the mac Telephone softphone with voip.ms. Didn't use it much after basic testing, as I wanted to get a sip desk phone, plus figure out the automated attendant (that's still in the future). Today, went to install a sip phone and found that...
  18. S

    SOLVED AnveoDirect on Xivo - Incoming calls busy

    I have a recent install of Xivo that I setup with Google Voice (via Simonics) and AnveoDirect. GV works flawlessly as does outbound calling on Anveo. However, when I dial into the Anveo number I get a busy signal and this line in the logfile: NOTICE[1750][C-00000010] chan_sip.c: Call from...
  19. EcstaticMark

    QUESTION Caller Not Getting My 200 OK

    I'm losing sleep over this, so here goes. Short version: I can't answer inbound calls to my trunk's DIDs, but CAN answer calls on the same trunk to my ported DID. Asterisk Ver. 13.6.0, Incredible PBX 12.0.74. 1 Digium Trunk with 2 lines. 234-521-xxx1 and 234-521-xxx2. One 1 line, my home number...
  20. ChiefGyk

    NO JOY Avantfax not letting me log in

    So I set up a fresh CentOS VPS with IncrediblePBX, but Avantfax is not letting me log in on any browser with the new password I made. CentOS 6.7 VPS on Digital Ocean w/ IncrediblePBX 13
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