search for: livewirenet

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable.. registrations were ok, etc.. but now in head it's borked. verbose = 30 debug = 30 sip debug on.. *CLI> <-- SIP read from 66.185.98.152:5060: REGISTER sip:voi...

...seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend accountcode=asteriskgw host=x.x.x.x restrictcid=no fromdomain=voip.livewirenet.com context=lwn canreinvite=yes Incoming calls from the peer lack an accountcode field.. a NoOp(${ACCOUNTCODE}) is blank for calls coming from the above peer.. forcing an accountcode does indeed work with SetAccount() however, as I stated, this does not scale. I should also note that both incom...

Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url :

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

Has anyone setup SIP services with ss7 and lis trunks? If so .. what was used hardware and software.. we're trying to do a SIP -> pstn setup and not having much luck as QWEST keeps pushing dates off (aka trying to screw us over) for our pri lines due to the recent court and fcc activity in regards to unbundled switching and I'm looking for solutions/ideas involving SS7..

I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get "you have" and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. -------------- next

Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. -------------- next part -------------- A non-text attachment was

We are receiving multiple audio drop outs on calls .. I've done quite a bit of troubleshooting and it only involves calls that require the Dial(SIP/xxx,,t) for transfers.. as long as the media path goes through the server the audio blips happen.. using ulaw codec, btw. I have been able to align the blips in audio to a specific point involving asterisk.. it seems to happen right at about

I'm looking for opinions on g726-32 vs. g711u.. They both have decent audio quality.. and looking at the wiki I get the impression that g726 is like the little brother to g711. Yet, I've run into quite a few sip termination vendors who don't support it. Does anyone on the list actively use g726 for anything and what have those experiences been? The g726 codec for me at least

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx