I have been struggling with several mediocre SIP FXO gateways on Asterisk for the past 6 months and have found that each one has at least one major problem with it. I am looking for any success stories using 1 to 4 port SIP FXO gateways on Asterisk. I need for them to support RFC2833 DTMF bridging each way and G729 codecs. Multi-port FXO gateways need to have some sort of grouping and/or routing of SIP calls to specific channels. So far, I have evaluated an Audiocodes MP-104 (4-port) and a Multitech MVP-130 (1 port). If anyone has found something that works in these scenario's, I would love to hear from you. I want to deploy many small FXO gateways over a large geographical area and would like to find something that actually works. Thanks for the help!
------------------------ From: Clif Jones <ctjones@earthlink.net> Subject: [Asterisk-Users] FXO gateways on Asterisk Date: Wed, 18 Feb 2004 00:00:00 -0500 To: asterisk users <asterisk-users@lists.digium.com>> I have been struggling with several mediocre SIP FXO gateways on > Asterisk for the past > 6 months and have found that each one has at least one major problem > with it. I am looking > for any success stories using 1 to 4 port SIP FXO gateways on Asterisk. > I need for them > to support RFC2833 DTMF bridging each way and G729 codecs. Multi-port > FXO gateways > need to have some sort of grouping and/or routing of SIP calls to > specific channels. So far, > I have evaluated an Audiocodes MP-104 (4-port) and a Multitech MVP-130 > (1 port). > If anyone has found something that works in these scenario's, I would > love to hear from you. > I want to deploy many small FXO gateways over a large geographical area > and would > like to find something that actually works. Thanks for the help!I'm somewhat in the same boat and have been evaluating the Mediatrix 1204 sip gateway. Although I've not tried these options to see if they actually work, the following suggests the box supports your stated requirements: Select preferred port #1 codec. 0 = G.711 u-Law (PCMU) 1 = G.711 a-Law (PCMA) 2 = G.723.1 3 = G.729.AB DESCRIPTION "Attribute to select the type of DTMF transport. 0 = In band 1 = Out-of-Band using Signaling Protocol 2 = Out-of-Band using RTP DTMF payload So far my reseller has been able to find ways to accomplish everything that I've bumped against in terms of implementation issues. The only remaining issue that I'm currently having is when C7960 -> asterisk -> Mediatrix g/w call is completed, when the C7960 user disconnects it sends the BYE to asterisk, but asterisk "intermitently" does not send a BYE to the Mediatrix. (Think you and I exchanged a couple of emails on this.) Since asterisk was Dec 4th CVS, I updated code this past weekend and still have the exact same problem. I need to research that further. As of this moment in time (might change tomorrow), the 1204 seems to be about the only 1-to-4 port sip g/w box on the market that is even close to being reasonable. Rather spendy for what its really doing though. Rich
Hi, I have tested, and still am testing different sip equipment with asterisk. In my case I'm testing fxs gateways but this is what I have found so far: - audiocodes mp-104 fxs sip: I'm still testing this one but so far I can not get it to register properly with asterisk, it fails to authenticate. Messages are exchanged properly but asterisk just fails to accept its proxy-authorization data, as if md5 hashed password was not accepted. If you comment out secret= in sip.conf it registers properly, but there is no authenticartion. Sometimes I get retransmission RTP errors (asterisk/syslog). RFC2833 dtmf - ok. I heard that it hungs... ?? Pitty cause it looks solid, works quietly..... - mediatrix 1104 fxs sip: Registers properly. RFC2833 dtmf signalling. G729 - ok. Very nice VAD mechanizm. Every now and then I got unhandled sip request errors (asterisk/syslog). Basically the only bad thing I can say about this box is that it's pretty loud. regards, Dave -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Clif Jones Sent: Wednesday, February 18, 2004 6:00 AM To: asterisk users Subject: [Asterisk-Users] FXO gateways on Asterisk I have been struggling with several mediocre SIP FXO gateways on Asterisk for the past 6 months and have found that each one has at least one major problem with it. I am looking for any success stories using 1 to 4 port SIP FXO gateways on Asterisk. I need for them to support RFC2833 DTMF bridging each way and G729 codecs. Multi-port FXO gateways need to have some sort of grouping and/or routing of SIP calls to specific channels. So far, I have evaluated an Audiocodes MP-104 (4-port) and a Multitech MVP-130 (1 port). If anyone has found something that works in these scenario's, I would love to hear from you. I want to deploy many small FXO gateways over a large geographical area and would like to find something that actually works. Thanks for the help! _______________________________________________ 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