search for: ipcomms

Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 N...

Dear Sirs, I?m writting a pam module for Radius authentication, which should allow a remote login via ssh on a Linux machine with an openssh server. In particular, the user which is configured at the remote Radius server is not present on the local user database of the Linux machine. Unfortunatly, openssh will not allow to start a PAM user authentication if cannot find the user login in the

...ience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant problems since the first day. Turns out (I found this out approx 1 month ago) that the "version of Asterisk" they are using is "Asterisk at Home 1.x" So basically is there any way to determine the version of Asterisk being used? I cant t...

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38

Hello How do I fix this.... [IPComms-in] exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN01},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN02},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCID...

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

...sip.myphoneprovider.com). If you restrict your inbound sip to that, it will be very helpful. Also, there are further steps you can take to harden your systems. An internet search will bring up many, but here are a couple of good ones: http://blogs.digium.com/2009/03/28/sip-security/ http://www.ipcomms.net/blog/70-11-steps-to-secure-your-asterisk-ip-pbx http://nerdvittles.com/?p=580 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150608/8f06dc72/attachment.html>

...I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.149 context=incomingcall disallow=all allow=gsm allow=alaw allow=ulaw in extensions.conf I have at the bottom of [incomingcall] : exten => 9546782688,1,AGI,calleridlookup.agi (script I wrote to rewrite callerid based on ph...

Hello All, Do you guys suggest any 1800 DID Provider in the US ? I'm having a hard time to find one. Thanks, Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091127/63d93945/attachment.htm

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host proxy.ideasip.com, port 5060

Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can.

Hi list! Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message: == Using SIP RTP CoS mark 5 -- Executing [000972592603325 at default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325 at default:2]

...many Providers... > inbound sip to that, it will be very helpful. Also, there are further > steps you can take to harden your systems. An internet search will bring > up many, but here are a couple of good ones: > > http://blogs.digium.com/2009/03/28/sip-security/ > http://www.ipcomms.net/blog/70-11-steps-to-secure-your-asterisk-ip-pbx > http://nerdvittles.com/?p=580 OK, I set alwaysauthreject = yes and I discovered a allowguest, which I set to "no", too. The PBX is behind a Firewall and I just allow UDP 5060 and 10000-10100. Now I log the SIP-pakets coming from In...

All, I've put a new asterisk server at another location and can't seem to get it working. What's the best strategy to debug connections? I'm doing inbound SIP only and have installed the server in the same way as I did on my DEV server. Running an nmap on localhost shows the port listening: ---------- [asterisk]/> nmap -sU localhost Starting Nmap 4.11 (

Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short- term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They