similar to: Force codec order

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

Hi, someone can help me, I am ot able to find the way how to user Ajax.updaterto test if the request give some positive or negative result. I am able only to return the result inside a div. An example is appreciated. _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org

Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me?

Hi, someone know a good webphone, possibily a free one Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm

Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070209/6780fde6/attachment.htm

Hi, I saw one site (bubbleshare) that it is able to caputer the click on the log in link, however, I cannot understand how they can do that Someone can explaint it to me? Thank you _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org http://lists.rubyonrails.org/mailman/listinfo/rails-spinoffs

Hi, I'm using a script in order to send out my faxes with the application txfax, therefore, I do not know how to see if the faxes are sent. Any idea?

I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register => user:pass@provider/400 [inside] exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT) exten => _4X.,2,hangup -- Executing

hi using the following in sip.conf, codec preferences aren't set, and asterisk uses alaw whatever I do, except force it to one specific in the [user] [general] disallow=all allow=g726 allow=g729 allow=gsm allow=alaw then, from 'sip show peer something' it tells me Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (none) can someone please explaing why? this is

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):

Hi, I set a preference order of the codecs to my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls of not registered phones disallow = all allow = g729 allow = g723 allow = alaw allow = ulaw Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec. Problem: asterisk cannot make

Hi, It seams that ix.dovecot.fi has not build a new version since 2016-04-19 20:03 regards, -- Harald Leithner ITronic Wiedner Hauptstra?e 120/5.1, 1050 Wien, Austria Tel: +43-1-545 0 604 Mobil: +43-699-123 78 4 78 Mail: leithner at itronic.at | itronic.at

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I?m a PhD student at the University of Warsaw, and have started using R. In many books they specify to use <- instead of = when assigning values, and this is also mentioned in older posts on the R website. However, it seams to me that some update has occured, becuase I continously get the same result wether I use <- or =. I would be extremely helpful for any answer to this. = seams more

Hi John, I'm forwarding this email to the relevant developer lists at Xiph.Org, where the format was created and where the experts gather. I'm certain there is a design reason for putting the bos pages of all logical bitstreams together at the start. My guess is that with digital media you usually have a delay time for setting up the decoders for a specific format and by clustering

Hi John, I'm forwarding this email to the relevant developer lists at Xiph.Org, where the format was created and where the experts gather. I'm certain there is a design reason for putting the bos pages of all logical bitstreams together at the start. My guess is that with digital media you usually have a delay time for setting up the decoders for a specific format and by clustering

[outgoing-fax] exten => _0XXXXXXXXX,1,SetVar(NumberCalled=${EXTEN}) exten => _0XXXXXXXXX,2,Wait(10) exten => fax,1,SetCallerid(${FAX_CALLERID}) exten => fax,2,Dial(Zap/g1/${NumberCalled},60) exten => fax,3,Hangup exten => t,1,Busy exten => i,1,Busy -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

Hi all samba coders. Glad to see that you work to improve samba. Here is a problem I would be very greatful if you could solve. Certain international characters does not work as usernames when trying to log in to an NT4 file server. My problem is that in Swedish versions of NT the "Administrator" username is translated and reads "Administrat?r". It seams that the

How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when