[Asterisk-Users] Using MultiTech MVP-210 as FXO/FXS gateway
Hi all, Im trying to use my 2-port multi-tech VoIP gateway to talk to asterisk. Ideally I want to put it in a remote location with a POTS line one port1 and an analog phone on port2 to call that location. Both the MultiTech and Asterisk have non-natted static IPs. I have tried every different type of configuration possible for the sip.conf file. I can call from the analog phone on the multitech to a local asterisk extension and it rings, but when I pickup I get a busy signal at both ends. When I try and call from asterisk to the phone on the multitech, I dont even get that far. I receive this from the CLI: -- Starting simple switch on 'Zap/10-1' -- Executing Dial(Zap/10-1, SIP/multitech) in new stack -- Called multitech -- Got SIP response 486 Busy Here back from 122.33.44.55 -- SIP/multitech-964c is busy == Everyone is busy at this time n Hungup 'Zap/10-1' The MultiTech seems pretty simple to configure, just the IP of asterisk, username and pass. The only field I havent tried its SIP URL. I was recently at a MultiTech show and I saw them use x-lite to call to the MultiTech. Since neither is a sip proxy, I cant figure out why that worked for them but I cant get this working with asterisk. Here is the current version of my sip.conf [multitech] context=local ;disallow=all allow=all ;disallow=all allow=gsm allow=ulaw allow=alaw type=friend username=multitech secret=pass nat=no ;mailbox=200 host=dynamic reinvite=no ;canreinvite=yes qualify=1000 dtmfmode=info canreinvite=no callerid=Multi Tech ;defualtip=1.2.3.4 Thanks everyone, Steve
Re: [Asterisk-Users] Using MultiTech MVP-210 as FXO/FXS gateway
I tested Multitech with the same scenario and it works. Stephen Foster wrote: The MultiTech seems pretty simple to configure, just the IP of asterisk, username and pass. The only field I havent tried its SIP URL. I was recently at a MultiTech show and I saw them use x-lite to call to the MultiTech. Since neither is a sip proxy, I cant figure out why that worked for them but I cant get this working with asterisk. No so simple. At least you must to elaborate the following windows: IP, Voice/Fax, Interface, Phone Book configuration,Outbound Phone Book, Inbound Phone Book. Here is the current version of my sip.conf [multitech] context=local My sip.conf: [multitech] context=default type=friend host=192.168.YY.XX ; multitech IP dtmfmode=inband; we use alaw Hope this help. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using MultiTech MVP-210 as FXO/FXS gateway
I am using an MVP-210 as FXS -- I haven't tried FXO. Here's my sip.conf entry: [mvp-x303] type=friend host=192.168.1.93 username=303 dtmfmode=rfc2833 context=fs1 disallow=all allow=ulaw (Not sure if dtmfmode is correct.) Username must be an extension number that appears in the MVP210's inbound phone book. Here's an MVP210 outbound phone book entry to call x352 on the Asterisk server: Destination Pattern: 352 Total Digits: 3 IP Address: 192.168.1.94(the Asterisk server) Protocol Type: SIP Transport Protocol: UDP (MVP210 defaults to TCP) SIP Port Number: 5060 At 01:23 PM 3/11/2004 -0500, Stephen Foster wrote: Hi all, Im trying to use my 2-port multi-tech VoIP gateway to talk to asterisk. Ideally I want to put it in a remote location with a POTS line one port1 and an analog phone on port2 to call that location. Both the MultiTech and Asterisk have non-natted static IPs. I have tried every different type of configuration possible for the sip.conf file. I can call from the analog phone on the multitech to a local asterisk extension and it rings, but when I pickup I get a busy signal at both ends. When I try and call from asterisk to the phone on the multitech, I dont even get that far. I receive this from the CLI: -- Starting simple switch on 'Zap/10-1' -- Executing Dial(Zap/10-1, SIP/multitech) in new stack -- Called multitech -- Got SIP response 486 Busy Here back from 122.33.44.55 -- SIP/multitech-964c is busy == Everyone is busy at this time n Hungup 'Zap/10-1' The MultiTech seems pretty simple to configure, just the IP of asterisk, username and pass. The only field I havent tried its SIP URL. I was recently at a MultiTech show and I saw them use x-lite to call to the MultiTech. Since neither is a sip proxy, I cant figure out why that worked for them but I cant get this working with asterisk. Here is the current version of my sip.conf [multitech] context=local ;disallow=all allow=all ;disallow=all allow=gsm allow=ulaw allow=alaw type=friend username=multitech secret=pass nat=no ;mailbox=200 host=dynamic reinvite=no ;canreinvite=yes qualify=1000 dtmfmode=info canreinvite=no callerid=Multi Tech ;defualtip=1.2.3.4 Thanks everyone, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
