On Mon, Dec 19, 2005 at 06:36:16AM -0600, Rich Adamson wrote: > ok rick i will check this directives and write you again.. thanks!!!! > > ok rick all of my conf... > > asterisk 1.2.1 > > zaptel 1.2.1 > > > > i have a pbx simple with digital phones in one side. and the other side > > are xten with SIP. > > > > my extencion.conf > > [general] > > static=yes > > writeprotect=no > > autofallthrough=yes > > > > [globals] > > CONSOLE=Console/dsp ; Console interface for > > demo > > TRUNK=Zap/g1 > > [local] > > ; ignorepat => 9 > > include => default > > > > [default] > > ; > > ; By default we include the demo. In a production system, you > > ; probably don't want to have the demo there. > > > > exten => 402,1,Dial(SIP/402,20) > > exten => 402,2,Hangup > > > > [teste] > > exten => s,1,Dial(SIP/402,20) > > exten => s,2,Hangup > > exten => 402,1,Dial(SIP/402,20) > > exten => 402,2,Hangup > > > > exten => _XXX,1,Dial(${TRUNK}/${EXTEN}) > > exten => _XXX,2,Voicemail(u${EXTEN}) > > > > > > > > the sip.conf is the default for asterisk i didnt touch anything in this > > file only the extention number and i dont have nothing about codecs in > > this file > > > > [402] > > type=friend > > host=dynamic > > username=Pablo > > secret=teste > > callerid="Pablo" <402> > > canreinvite=no > > ;nat=yes > > ;amaflags=billing > > context=teste > > > > > > > > > > > > Hi all i have some problems with my pbx and asterisk codecs. > > > > > > > > > > > > if i use g711u or g711a codecs. the line never hangup. and the > > > > > > origin > > > > > > and destination are connected until i restart my pbx or asterisk > > > > > > > > > > > > But if i use GSM all work fine. > > > > > > > > > > > > is possible to solve this problem? or use only gsm codec? > > > > > > > > > > > > > > Yes, its possible to solve the problem. > > > > > > > > can you explain how? > > > > > > Not without you providing at least "something" to give us a clue what it > > > is that you've programmed into your system. > > > > > > How about if you give us some clue as to which version of * you're > > > using, what type of phones are associated with "origin" and "destination", > > > if these are sip phones what do your sip.conf definitions look like, > > > what does the appropriate sections of extensions.conf look like, and > > > any other configuration pieces that might pertain to whatever it is > > > that you've implemented. Your posting implies there might be more than > > > one * system involved and possibly even iax trunking, etc. > > Okay, start with "show translation" to see which codecs you system > can translate. > > Then check your sip phones to see which codecs are supported. For the xten > product (as with most sip phones), you can select which codecs to support > and which ones are preferred. > > In sip.conf you are only showing one sip phone. Are there more defined > that you didn't paste into this email? Based on the data that you've > provided, you only have one phone and its on extension 402. Since there > is nothing else defined (at least based on your config files), you > won't be able to call anyone. > > You can control which codecs are used by doing something like this: > [402] > type=friend > host=dynamic > username=Pablo > secret=teste > callerid="Pablo" <402> > canreinvite=no > disallow=all > allow=ulaw > context=teste > mailbox=402 > > [403] > type=friend > host=dynamic > username=Pablo2 > secret=teste2 > callerid="Pablo" <403> > canreinvite=no > disallow=all > allow=ulaw > context=teste > mailbox=403 > > Later on when you want to start playing with voicemail, you will want to > add the statement shown above (mailbox=402). > > In extensions.conf, you need entries like this: > [teste] > exten => 402,1,Dial(SIP/402,15) > exten => 402,2,Voicemail(u402) > exten => 402,102,Voicemail(b402) > exten => 402,103,Hangup > > exten => 403,1,Dial(SIP/403,15) > exten => 403,2,Voicemail(u403) > exten => 403,102,Voicemail(b403) > exten => 403,103,Hangup > > With the above, extension 402 can call 403 as well as 403 can call 402. > > Your entry > exten => s,1,Dial(SIP/402,20) > exten => s,2,Hangup > does not apply to the configuration that you've shown us. The "s" extension > is typically used for calls that arrive via Zap and Iax channels where > "no dialed digits" are received. The "s" is not a match-all option. > > We don't have any idea what you mean by "the other side". If you are > trying to dial from one sip phone to another on your system, then you > need to define each phone in sip.conf as shown above, and configure > each phone so that it properly registers with asterisk. To see what > is registered, do a "sip show peers". If you sip phones don't show in > that list, they aren't registered. Fix that first before moving on. > > Once the above configs have been fixed and asterisk restarted, then > watch the asterisk CLI to "see" what happens when one phone calls the > other. If you still have problems, paste the CLI output into a posting > for us to see. Without that, we can only guess. > > Given what you have posted, I don't have a clue what you are trying to > do with: > exten => _XXX,1,Dial(${TRUNK}/${EXTEN}) > exten => _XXX,2,Voicemail(u${EXTEN}) > However, when sip extension 402 dials 403, it will match the above _XXX > and send that call out Zap/g1 (whatever that happens to be). > > If you really are working with two asterisk systems tied together with > Zap channels, then I'd suggest modifying the above to something like > exten => _5XX,1,Dial(${TRUNK}/${EXTEN}) > exten => _5XX,2,Voicemail(u${EXTEN}) > when the 4XX extensions are on one system and the 5XX extensions on the > second system. > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text---
-- .- Pablo Allietti LACNIC _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users