similar to: 2 problems I can't solve without any help

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Hello list. My phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

I received an e-mail from a vendor who says: "We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP." I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what

It seems to me that ITSP's like to use a US dialing code eg 1-xxx Wouldn't it be cool to have an Internet dialing code?? I don't know what the structures are or how the allocations work but it would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx was an internet phone.. That way the whole internet phone space could be consolidated into a single dialing structure

Interesting read in Alec Saunders blog today. http://saunderslog.com/2007/03/01/mashable-telcos/ Thought it may interest some people on this list. As food for thought, why it is that ITSP's haven't come up with more 'interesting' voice applications? When asterisk first became available I thought it was the beginning of seeing really neat applications, think Verzion's

With the recent service outage at Broadvoice, there has been a lot of discussion here, on broadband reports, Voxilla, etc., regarding whether VOIP is mature, or "ready for the masses", etc. One particular point I've seen repeated, and with which I agree: "we're willing to deal with less than five 9s, even one or 2 9s, as long as we have good communication regarding the

Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. asterisk*CLI> sip show registry Host Username Refresh State Reg.Time 85.119.188.3:5060 092779077 105 Failed Sun, 05 Jul 2009 23:11:40 asterisk*CLI> sip reload [Jul 6 10:30:43]

I have been able to setup three different providers successfully, but only one at a time. I would like to have all active in a fail over configuration so that one failing would not be noticed by the users. I know it's probably easy to configure but I have not been able to find out how. Can anyone give me an example? Chris Mason

All, I use various IAX providers to terminate my outbound calls. I set the caller-ID to one of several DIDs, based on the called number. There doesn't seem to be any rhyme or reason as to what the called user sees, however. Calls to most cell-phones show *exactly* the number I submit. Calls to land-lines sometimes show, other times not. Calls to other voip-providers most often show, but

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco.

If I connect to a provider using iax, and that provider connects to his provider using only sip, the provider I am connecting to isn't going to be able to bridge the call and drop out of the media stream correct? If I'm understanding how bridging works, you lose the ability to have the media stream going directly between the two endpoints of the call with most of the providers out there