[asterisk-users] AddQueueMember() and PersistentMembers

Hi, I'm trying to use AddQueueMember() to add a member to a queue and trying to make this logged member in the queue between reloads and restarts of asterisk. I configure en queues.conf: [general] Persistentmembers=yes And Extensions.conf: exten=

[Asterisk-Users] Dial Limit Call Options

Hi, Is there a way to know if after using the Dial command and specifying L(X:Y:Z) option for limiting the duration of the call and if the calls reachs that limit have an indication that the caller reachs the limit? (i.e. DIALSTATUS) Thanks Alejandro Ghergherian

RE: [Asterisk-Users] Clicks in audio with TE100P PRI

? == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Alejandro G wrote: I tested all again

[Asterisk-Users] ADSI over SIP

Hi, Does anybody knows if ADSI could be used from the SIP channel? Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Clicks in audio with TE100P PRI

It seems that configuring span=1,1,0,ccs,hdb3 and changing jitterbuffer=16 resolves or masks the issue. What I will do now is reduce again jitterbuffer to default to see what happens. To answer some of the questions I don't see hard disk activity when the clicks appear, also the hard disk has

[Asterisk-Users] Clicks in audio with TE100P PRI

I tested all again. No matter if span=1,1,0 or span=1,0,0 if I configure jitterbufer=4 I have glitches that I'm almost sure that are holes in audio. If I raise jitterbufer=16 the problem disappear (or becames impercetible). Anyway I am interested in understand what is happening. Your issue

[Asterisk-Users] Clicks in audio with TE100P PRI

modify it? Thanks, Alejandro G. ___ 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

[Asterisk-Users] Clicks in audio with TE100P PRI

Thanks for your answer. Googling in the lists I found what you are telling that maybe there is a synchro problem with the E1, but I'm not so sure that this could be. I am configuring zaptel.conf like this: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 But I also changed to test to:

[Asterisk-Users] DTMF callerid does work

Hi, I am trying to make work dtmf callerid with X100P with no success. In the case I am working we receive the DTMF before the ring and/or polarity inversion and nothing happens (I understand that X100P do not recognize polarity inversion). We start ooking at bug 9 and bug 1719 and found some

[Asterisk-Users] CID signalling for DTMF

in this condition? Thanks, Alejandro G. ___ 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

[Asterisk-Users] Does ztmonitor record the audio channel?

is allways 0 with no data. Is anything I am doing wrong? Any hint about using ztmonitor? Thanks Alejandro G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] cidsignailling mode question

Hi, I need to use cidsignalling=dtmf where the callerid comes after the first ring. Looking in source code of chan_zap.c I understand that cidsignalling=dtmf only works when cidstart=polarity. Is this right? or also works with cidstart=ring? Thanks Alejandro

[Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression

Hi, I have a problem with ATA-186 configured for silence supression (AudioMode bit 0 = 1). When enabled and listening music on hold no sound is heared (if I talk I began to hear the music and again mutes when I stop talking). If I configure for silence supression off everything goes fine. Any

[Asterisk-Users] Group channel rotation for outgoing call?

Hi, If I have a PRI with all channels grouped in group=1, I understand when I want to make an outgoing call that asterisk takes the first channel available. Is there any possiblity to rotate the channel taken? I was searching in Wiki but I could not find nothing about. Thanks, Alejandro

[Asterisk-Users] Help in E1-T1 encoding

Peter, Sorry for the delay. I had to reconfigure all again. I do I inbound call to asterisk and the result log is this (hope this is usefull): Alejandro Enabled EXTENSIVE debugging on span 1 *CLI T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (27) [ 00

[Asterisk-Users] Help in E1-T1 encoding

Peter, Should I do the with pri intense debug span or pri debug span only? I will need a little more time for install again the T1's. Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Help in E1-T1 encoding

The call is received from the PSTN by an NMS E1 with ISDN Pri (EuroISDN) and is switched to the NMS T1 using NMS MVIP internal bus switch that is similar to a H.100 bus with few streams/timeslots. Once in the T1 (which is running National ISDN 2 Pri) this board calls the TE110P in asterisk

[Asterisk-Users] Help in E1-T1 encoding

? Thanks in advance. Alejandro G ___ 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