I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better on slow connections? Jitter more/less of a problem then w/ g.711? Was implementation a pain? I've seen the bandwidth comparisons @ http://www.voip-info.org/wiki-Bandwidth+consumption Things look good... if g.729 turns out to be all it perports itself to be then I feel we'd have a real winner.
> I have a few SIP phones, Cisco 7960s, and was looking into implementing > some compression, ala G.729. I'm looking into purchasing a g729 > licenses just to get an idea of performance and voice quality, over > lans, wireless and single channel isdn. > > Does anyone have positive/negative experience w/ getting > licenses/support from Digium? Hows the sound quality compared w/ > g.711? Is 729 better on slow connections? Jitter more/less of a > problem then w/ g.711? Was implementation a pain? I've seen the > bandwidth comparisons @ > > http://www.voip-info.org/wiki-Bandwidth+consumption > > Things look good... if g.729 turns out to be all it perports itself to > be then I feel we'd have a real winner.We've got about five licenses and a remote 7960's v6.3 running over dsl working just fine. The average user cannot tell the difference between 711 and 729. Installation was easy and straight forward, although you'll find comments in the archives that 729 installation requires a non-scsi drive on the * box. In some cases, you might require two licenses even though you might have only a single 729 phone. Think about VM, etc. Error on the side of too many. Can't comment on support; never needed any.
Hi, I have setup and configured asterisk server using SIP. I defined two users 2000 and 2001 in the sip.conf file. The extension defintions from sip.conf are shown below: [2000] type=friend username=2000 secret=hi host=dynamic context=from-sip mailbox=100 [2001] type=friend username=2001 secret=hi host=dynamic context=from-sip mailbox=101 I made entries in extensions.conf, shown below, to place calls to 2000 and 2001 [from-sip] exten=>2000,1,Dial(SIP/2000,20) exten=>2000,2,Voicemail(u2000) exten=>2000,102,Voicemail(b2000) exten=>2000,103,Hangup exten=>2001,1,Dial(SIP/2001,20) exten=>2001,2,Voicemail(u2001) exten=>2001,102,Voicemail(b2001) exten=>2001,103,Hangup Initially I tested out the server by registering SJPhone user agents and successfully placing calls between them. Next I replaced the SJPhones with our VOIP gateways. Everytime I dialed either extension I always got the unavailable IVR message. I tried looking deeper into the problem and took ethereal traces and was able to isolate the problem. For some reason asterisk has problems in rewriting the TO header field when it forwards the INVITE request to the callee. This is what the TO header field looks like when it is sent by the caller to asterisk (192.168.0.44 is the IP address of asterisk): To: <sip:2000@192.168.0.44:5060> and this is what it looks like when it is forwarded to the callee by asterisk (192.168.0.243 is the IP address of the callee). To: <sip:192.168.0.243> Since the URI does not contain the user part the callee replies with 404 not found and the call fails. I have thought hard, compared signaling traces but cant really make out how to make my gateways work, seems like an asterisk bug. Any ideas? I would really appreciate any help in this regard. Regards, Danish
you also have fromuser fromdomain use em bkw ----- Original Message ----- From: "usmankhan" <usmankhan@iphonica.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, May 15, 2004 6:10 AM Subject: [Asterisk-Users] TO header field bug using asterisk> Hi, > > I have setup and configured asterisk server using SIP. I defined > two users 2000 and 2001 in the sip.conf file. The extension > defintions from sip.conf are shown below: > > [2000] > type=friend > username=2000 > secret=hi > host=dynamic > context=from-sip > mailbox=100 > > [2001] > type=friend > username=2001 > secret=hi > host=dynamic > context=from-sip > mailbox=101 > > I made entries in extensions.conf, shown below, to place calls to > 2000 and 2001 > > [from-sip] > exten=>2000,1,Dial(SIP/2000,20) > exten=>2000,2,Voicemail(u2000) > exten=>2000,102,Voicemail(b2000) > exten=>2000,103,Hangup > > exten=>2001,1,Dial(SIP/2001,20) > exten=>2001,2,Voicemail(u2001) > exten=>2001,102,Voicemail(b2001) > exten=>2001,103,Hangup > > Initially I tested out the server by registering SJPhone user > agents and successfully placing calls between them. > > Next I replaced the SJPhones with our VOIP gateways. Everytime I dialed > either extension I always got the unavailable IVR message. I tried looking > deeper into the problem and took ethereal traces and was able to > isolate the problem. For some reason asterisk has problems in > rewriting the TO header field when it forwards the INVITE request to > the callee. This is what the TO header field looks like when it is > sent by the caller to asterisk > (192.168.0.44 is the IP address of asterisk): > > To: <sip:2000@192.168.0.44:5060> > > and this is what it looks like when it is forwarded to the callee by > asterisk (192.168.0.243 is the IP address of the callee). > > To: <sip:192.168.0.243> > > Since the URI does not contain the user part the callee replies with > 404 not found and the call fails. I have thought hard, compared > signaling traces but cant really make out how to make my gateways > work, seems like an asterisk bug. Any ideas? I would really appreciate any > help in this regard. > > Regards, > Danish > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >