[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=
*01,1,AddQueueMember(queue_name,Local/${CALLERID(num)[EMAIL PROTECTED],penalty);
When I log with AddQueueMember to any queue and stop and load asterisk
again, the database entry disappear. Is this a normal behavior?
I tried to look at the code in app_queue.c and check at
reload_queue_member() function, that function does not found the database
entry /Queues/PersistentMembers/queue_name.
Am I wrong? Any help?
Thanks.
Alejandro Guercio
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[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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Clicks in audio with TE100P PRI
In zapata.conf ; Configure jitter buffers in zapata (each one is 20ms, default is 4) ; jitterbuffers=16 Alejandro Ghergherian -Mensaje original- De: Rod Bacon [mailto:[EMAIL PROTECTED] Enviado el: Domingo, 25 de Septiembre de 2005 08:32 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion CC: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Clicks in audio with TE100P PRI Which file does the jitterbuffer setting go in, zaptel or zapata.conf? I can't find it documented anywhere. What version of zaptel drivers include a jitterbuffer? == 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. 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 is very likely the size of the zaptel jitterbuffers setting. If the zaptel driver is not immediately available to accept a frame of data it places it in an internal queue of pending writes. If that queue is full then the write is refused by the zaptel layer and then silently discarded by chan_zap causing a gap in the audio once it is played out of the zaptel card. If you crank up the debug level you will probably see 'Write returned -1...' (aka. EAGAIN) debugs that mostly correlate to the pops and clicks. Note that the zaptel driver legitimatly (if perhaps not appropriately) also refuses data when the channel is muted, such as during DTMF generation and at other times, so not _all_ EAGAIN debugs are a sign of problems. This makes perfect sense but again some issues of the problem do not match. I set debug at level 9 and there is no message of errors. Another thing I do not understand is why the same configuration: PAP2 - LAN - Asterisk - TE100P works perfect, and instead of LAN using internet generates the problem. Shouldn't it be the same for both configs? The only difference I see is that the rtp packets came from another Ethernet card, but if I call to terminate calls with another carrier using that eth works fine. What is clear is that jitterbuffer=16 corrects the problem. One more thing: no matter what codec I use, G729 or G711 the sound clicks are almost the same. Is anyway I could debug at RTP level in asterisk to see what is happening and check if there is packet loose? Thanks Alejandro ___ 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 very low usage. The clicks I listened were continuous and periodic. If the other party stays in silence I also listen the click every half second. Also to check, I run zttest and gives me Best=100%, average=99.989%. Once tested again I'll write the results to see what happen. Thanks to all Alejandro ___ 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
[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 is very likely the size of the zaptel jitterbuffers setting. If the zaptel driver is not immediately available to accept a frame of data it places it in an internal queue of pending writes. If that queue is full then the write is refused by the zaptel layer and then silently discarded by chan_zap causing a gap in the audio once it is played out of the zaptel card. If you crank up the debug level you will probably see 'Write returned -1...' (aka. EAGAIN) debugs that mostly correlate to the pops and clicks. Note that the zaptel driver legitimatly (if perhaps not appropriately) also refuses data when the channel is muted, such as during DTMF generation and at other times, so not _all_ EAGAIN debugs are a sign of problems. This makes perfect sense but again some issues of the problem do not match. I set debug at level 9 and there is no message of errors. Another thing I do not understand is why the same configuration: PAP2 - LAN - Asterisk - TE100P works perfect, and instead of LAN using internet generates the problem. Shouldn't it be the same for both configs? The only difference I see is that the rtp packets came from another Ethernet card, but if I call to terminate calls with another carrier using that eth works fine. What is clear is that jitterbuffer=16 corrects the problem. One more thing: no matter what codec I use, G729 or G711 the sound clicks are almost the same. Is anyway I could debug at RTP level in asterisk to see what is happening and check if there is packet loose? Thanks Alejandro ___ 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
[Asterisk-Users] Clicks in audio with TE100P PRI
Hi, I have a problem I will describe. I have PAP2 connected to the internet to an asterisk box with 2 TDM cards, one TE100P E1 with PRI and one TDM400P with 2 FXS an one FXO. When I call to the TDM400 cards from the PAP2 eveything is OK, sound quality is perfect. When I call to terminate the call in PSTN through E100P I hear clicks which aparently are RTP packet looses. This clicks are only heard in the PSTN side, not in the PAP2. If I connect PAP2 in LAN to the *, everything sounds is normal. So I evaluate the following: 1. Delay or something similar in internet could not be the problem because it works with TDM400P (same configuration) 2. The PAP2 could not be the problem because it works with TDM400 (and other ip phones) and in a LAN. 3. The TE100P could not be the problem because it works fine if the PAP2 is connected via lan and not via internet. 4. With other IP phones everything works fine. It seems that the combination of PAP2 - Internet - TE100P is the problem. Any suggestions? is there any jitter buffer adjust for the sip channel or zap in the * side only for the TE100P? I look that in zapata.conf there is a jitterbuffer parameters which defaults to 4, should I 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-users
[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: span=1,1,0,ccs,hdb3 The same thing happens. You may consider also that if I connect PAP2 to LAN everything works, also if I use other ip phone from internet works fine. I also check if I'm loosing interrupts and everything seems ok. Also I pull out the TDM400 from the box. At last I change jitterbuffer=16 and it works better, the clicks are reduced. Could this be possible? What is the function of this parameter in zapata.conf? I should tell you that the TE100P is connected to another E1 board (not a live E1) from Natural Microsystems which acts as a gateway to PSTN. This board works as a PRI master but I don't think that this could be the problem as long as using other phones or in LAN it works perfectly and the voice is clear with no clicks o sound looses. Thanks again, Alejandro ___ 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
[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 patches that I apply. This patches did not work so I modify wcfxo.c to make the detection of the DTMF start (or the noise that detects) more sensible with the result that it begans to detect something. This detection in some cases was perfect and in other cases did not detect anything at all. The results are random. Sometimes retrieves 2 cid digits (out of 10) and sometimes all. We record with ztmonitor the signal received and we found that everything appears fine. The cid starts with a 'D' then comes cid digits and ends with a 'C'. Every digits has an on time off 70ms and and off time of 70ms also. Any hint or anyone with experience about? Thanks. AlejandroG ___ 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
[Asterisk-Users] CID signalling for DTMF
Hi, Trying to make work cid in analog lines where DTMF signalling is used for ANI, I found an implementation where the ANI information comes directly before ringing the phone and without any polarity change of the line. Do anyome know if there is any way to make asterisk work receiving ani 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-users
[Asterisk-Users] Does ztmonitor record the audio channel?
Hi, I'm trying to debug cid information coming from an analog PSTN line using ztmonitor to record the call using the following command line: ./ztmonitor 1 -v -f call.raw This works fine, I see visually the audio rx and tx bar, the file call.raw was created but the length of this file 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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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
[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 hint? Anybody with same problem? Thanks. Alejandro ___ 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
[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 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
[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 01 01 37 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 027 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 37 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 027 P/F: 1 0 bytes of data -- ACKing all packets from 26 to (but not including) 27 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter [ 02 01 36 36 08 02 05 00 05 04 03 80 90 a2 18 03 a9 83 81 6c 0c 00 80 31 31 34 37 38 35 33 38 37 37 70 0c 80 30 31 31 35 35 34 34 35 34 30 39 7d 02 91 81 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 027 0: 0 N(R): 027 P: 0 47 bytes of data -- ACKing all packets from 26 to (but not including) 27 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 1280/0x500) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 00 80 31 31 34 37 38 35 33 38 37 37] Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) '1147853877' ] [70 0c 80 30 31 31 35 35 34 34 35 34 30 39] Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '01155335410' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) Sending Receiver Ready (28) [ 02 01 01 38 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 028 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ 00 01 36 38 08 02 85 00 02 18 03 a9 83 81 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 027 0: 0 N(R): 028 P: 0 10 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 1280/0x500) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Accepting call from '1147853877' to '01155335410' on channel 0/1, span 1 -- Executing Ringing(Zap/1-1, ) in new stack [ 00 01 38 38 08 02 85 00 01 1e 02 81 88 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 028 0: 0 N(R): 028 P: 0 9 bytes of data T_200 timer already going (2) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1280/0x500) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Executing Wait(Zap/1-1, 4) in new stack [ 00 01 01 38 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 028 P/F: 0 0 bytes of data -- ACKing all packets from 26 to (but not including) 28 -- ACKing packet 27, new txqueue is 28 (-1 means empty) -- Something left to transmit (28), restarting T200 counter [ 00 01 01 3a ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 029 P/F: 0 0 bytes of data -- ACKing all packets from 27 to (but not including) 29 -- ACKing packet 28, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 counter -- Executing Goto(Zap/1-1, callthrough|s|1) in new stack -- Goto (callthrough,s,1) --
[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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 which is connected by a loopback cable. Asterisk receives call, asnwer it and do the stuff planned in the dialplan. Both T1 boards where first configured using ulaw and then alaw. In both cases the sound had distortion. Today I test another configuration trying to isolate error. Instead of using a NMS T1, I used another NMS E1 and I configure TE110P as an E1 using in both boards also EuroISDN and everything began to work fine. I also connect the TE110P to a live E1 and again everything is Ok. I not sure that the distortion is an encoding difference but it really seems that there is the problem. Configuration for T1 in zaptel.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 alaw=1-23 I check with command in CLI zap show channel 1 the Default law indicates alaw as expected. If I comment last line works the same but it indicates ulaw. zapata.conf: callprogress=no signalling=pri_cpe switchtype=national overlapdial=yes usecallerid=yes hidecallerid=no immediate=no echocancel=no echocancelwhenbridged=no callerid=asreceived context=incoming-call channel=1-23 language=es rxgain=0.0 txgain=0.0 Sorry for so long mail and thanks. Alejandro Ghergherian ___ 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
[Asterisk-Users] Help in E1-T1 encoding
I have an asterisk with a TE110P configured as T1 which is behind a PSTN gateway. This gateway has an E1 to PSTN and a T1 to asterisk. This T1 is configured as Network and * as CPE. Every call I receive in E1 gateway is directly switched to asterisk using T1. Remember E1 is alaw. Both E1 and T1 have Natural Microsystems boards with a very simple software. When I call to E1 asterisk signalling works fine, receives the call, answers it and playback a messaage. This message is reproduced with distortion like when you have an encoding ulaw/alaw error. I tried to configure * T1 in alaw configuring it in zaptel.conf but the same happens. If instead of calling from E1 PSTN I use any SIP phone, the message is normally reproduced without distortion. I could not connect * directly to a T1 to test it because I do not have the possibilty to have installed one in Argentina. Any ideas? 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
