Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the offending line looks like this: [202] type=friend username=202 secret=voip-analog0 host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 It works fine when calling between internally, or when the SPA-2000 is the calling source, but if a call comes in on a zap channel, the one way audio problem appears. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us
You have problems with pgsql. Check it. Regards, Gus ----- Original Message ----- From: "Hekuran Doli" <asterisk@ati-kos.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, June 20, 2004 5:27 PM Subject: [Asterisk-Users] Midifyed-Prepaid-Application> Hello. > > I have compile asterisk with modifyed prepaid application and populated > the database to! I have fill the card, cardtype, cid, country, > countrycode, reselers. I have make a cid=22 and I have add a user with > username and callerid 22. But I allways get prepaid-no-aaa. Any one could > help me how to authenticate? > Note: I want to bill my local clients registred to my asterisk box usingsip.> > Best Regards > Hekuran > > > _______________________________________________ > 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
Hello. I have compile asterisk with modifyed prepaid application and populated the database to! I have fill the card, cardtype, cid, country, countrycode, reselers. I have make a cid=22 and I have add a user with username and callerid 22. But I allways get prepaid-no-aaa. Any one could help me how to authenticate? Note: I want to bill my local clients registred to my asterisk box using sip. Best Regards Hekuran
Upgrade your firmware on the SPA-2000 and see if it fixes the one way audio problem. I had this problem and worked with Sipura to get it resolved. If you are running a firmware earlier then version 2.0.6(c) then you will have this problem. Matt -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Seth Mattinen Sent: Sunday, June 20, 2004 2:00 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] One way audio Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the offending line looks like this: [202] type=friend username=202 secret=voip-analog0 host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 It works fine when calling between internally, or when the SPA-2000 is the calling source, but if a call comes in on a zap channel, the one way audio problem appears. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us _______________________________________________ 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
Morning all, I want to know if someone from you did have (or have) this problem: from time to time, doesn't matter which asterisk version I use (I mean I had stable and CVS on this computer during the last 9 monthes and today a CVS from last week is running), called party have no audio. This happend only with SIP. I'm using G729 from digium on a SMP machine, kernel-2.4.26. Any hint welcome. -- Daniel
Wilson Pickett a ?crit :>>What I face is that a SIP call to our GW has from time to time the >>behaviour to "loose" audio. Hanging up and retrying can work, but mostly >>we wait or use an IAX GW and try again and then it work. Can also take >>few hours before it work again. >> >> > >What RTP ports are used in asterisk and do the match those of the phones? > >Asterisk: rtpstart 6970 rtpend 7170 ATA186: RTP 5004 Remember that I face this problem from time to time only -- Daniel
Hi, I have a Budgetone 100, and a X-Lite soft phone, both registered with Asterisk. when a call is made between them (regardless of which way) the Soft phone can hear both ways, but the Budgetone can't hear the soft phone. Any ideas? -- Richard
Hi, I have a Budgetone 100, and a X-Lite soft phone, both registered with Asterisk. when a call is made between them (regardless of which way) the Soft phone can hear the Budgetone, but the Budgetone can't hear the soft phone. Any ideas? -- Richard
Hi, My configuration is iaxcomm<-->*(Sangoma 102)<-->pri. About 5% of iaxcomm--->pri calls loose iaxcomm-->pri direction audio during conversation. Any idea? Thanks. iax.conf --------------------------------------------------- [general] port=5036 disallow=all allow=alaw jitterbuffer=yes maxjitterbuffer=500 maxexccessbuffer=50 tos=0x04 qualify=no [agent1] type=friend username=agent1 secret=agent1 context=agent host=dynamic notransfer=yes zapata.conf --------------------------------------------------- [channels] context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes callprogressdetect=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=4.0 txgain=-2.0 group=1 channel => 1-15 channel => 17-31 channel => 32-46 channel => 48-62 __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com
Hi Group, Just want to asked if some of you have experience 1 way audio. Currently I am using two asterisk box. One handles the prepaid platform, and the other one is for media gateway connection. I am using asterisk version 1.2.4 and a 4E1 digium card with echo cancellation. My interconnection between the two asterisk is IAX and sip for the clients. Would it be in the configuration of my zap channel which causes this problem. Some times I have to dial twice or three times before I can have a good call. Any documentation or suggestion that would help me understand this problem will be a great help. Regards, Leonimar __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
I would say NAT somewhere misconfigured. On 6/28/06, leonimar cape <leo_mac_ph@yahoo.com> wrote:> Hi Group, > > Just want to asked if some of you have experience 1 > way audio. Currently I am using two asterisk box. One > handles the prepaid platform, and the other one is for > media gateway connection. I am using asterisk version > 1.2.4 and a 4E1 digium card with echo cancellation. My > interconnection between the two asterisk is IAX and > sip for the clients. Would it be in the configuration > of my zap channel which causes this problem. Some > times I have to dial twice or three times before I can > have a good call. > > Any documentation or suggestion that would help me > understand this problem will be a great help. > > Regards, > > Leonimar > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --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 >