I tried G711 and GSM and in both cases call quality degraded when the softphone was conferencing more than 2 people (note: not a meetme room).
- Waldo On Dec 7, 2005, at 5:45 AM, xcel wrote: I did use IAX2 but sound quality wasn't that good which codec are you using with IAX2 ? *********** REPLY SEPARATOR ***********
On 12/6/2005 at 9:22 PM Alvaro Parres wrote: Why using SIP instead of IAX2 ??? Only a question becouse i always use IAX
On 12/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: Well... not so perfectly.
What I'm experiencing is that during certain call volumes, many calls go thru from box1 to box2. However, there are some cases where I get this message:
Dec 6 11:11:19 WARNING[203]: chan_sip.c:9525 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"5095551212" <sip:[EMAIL PROTECTED]>;tag=as3e387d65'
and the caller gets busy signal. However, other callers calling the same number get thru with no problems. Why is this?
Thanks, Waldo
On Dec 5, 2005, at 10:30 AM, Waldo Rubinstein wrote:
> This worked perfectly. > > Thanks, > Waldo > > On Dec 5, 2005, at 4:32 AM, xcel wrote: > >> >> Try this >> >> ___________________________________ >> 1st Machine sip.conf >> >> [box2] >> username=box1 >> type=friend >> host= 10.0.0.2 >> secret=***** >> >> in extensions.conf >> >> exten => _XXXXXX,1,Dial(SIP/box2/${EXTEN}) >> >> __________________________________ >> 2nd Machine sip.conf >> >> [box1] >> username=box2 >> type=friend >> host=10.0.0.1 >> secret=***** >> >> in extensions.conf >> exten => _XXXXX,1,Dial(SIP/box1/${EXTEN}) >> >> --xce >> >> >> *********** REPLY SEPARATOR *********** >> >> On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote: >> >>> I have 2 Asterisk servers running 1.2.0. One of them is a PSTN >>> gateway. Currently they are connected using IAX2. I wanted to play >>> with SIP. >>> >>> I setup a sip entry (type=friend) in the PSTN gateway box and a sip >>> entry (type=user) in the second box in order to send calls using SIP >>> to the second box. This works fine. However, when I setup the second >>> box as type=friend in order for it to be able to send calls back to >>> the gateway box, then calls no longer work from gateway box to the >>> second box. The reported error is: >>> >>> Dec 5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite: >>> Failed to authenticate on INVITE to '"2125551212" <sip: >>> [EMAIL PROTECTED]>;tag=as0698b1b9' >>> >>> In the gateway box, my sip.conf looks like this: >>> >>> [general] >>> allowguest=yes >>> autocreatepeer=no >>> >>> [secondbox] >>> type=friend >>> host= 10.0.0.2 >>> secret=mysecret >>> >>> In the second box, my sip.conf looks like this: >>> >>> [general] >>> allowguest=yes >>> autocreatepeer=no >>> >>> [secondbox] >>> type=user >>> host=10.0.0.1 >>> secret=mysecret >>> >>> Any ideas on how to correctly set this up? >>> >>> Thanks, >>> Waldo >>> _______________________________________________ >>> --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 >> >> >> >> _______________________________________________ >> --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 >
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