[Asterisk-Users] Txgain Rxgain
Hi, I currently have a TE210P with 2 E1 lines, one of them goes to the Telco which is fine and the other one goes to a Siemens HiPath 3750 PBX. The problem is that signal that the HiPath return is to HIGH and generates a lot of echo even when talking with a PAP2 on the same subnet, although when using the PAP2 to dial to a PSTN works fine. Well, doing some testing I found that setting RXGAIN=-12 and TXGAIN=-6 I eliminate the echo problems between the HIPATH and my SIP phones, but now the calls made to the PSTN are very low, is there a way to set RX TX gains diferently on each TE210P E1? Regards, Humberto ___ --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
[Asterisk-Users] Asterisk and Siemens HiPath 3750 issues
Hi, I'm currently facing some issues regarding echo between the asterisk box and the 3750, here is my scenario: TELCO -- Asterisk -- HiPath 3750 (E1) (TE210P) | SIP PHONES When I dial from a SIP phone to a Telco number it works fine, when I dial from a SIP to a HiPath extension I get too much echo. When dialing from the 3750 to the telco I also get echo. I have included my configuration file as well as other info. Has anyone had experience integrating asterisk with a 3750? Could anyone share the zaptel.conf e zapata.conf files? Thanks in advance, Humberto zaptel.conf # ISDN PRI - Telco span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=us defaultzone=us # HiPath 3750 span=2,2,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 loadzone=us defaultzone=us zapata.conf [channels] language=br facilityenable = yes jitterbuffers=8 switchtype=euroisdn pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes usecallingpres=yes hidecallerid=no callerid=asreceived musiconhold=default transfer=yes cancallforward=yes callreturn=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes rxgain=-1.0 txgain=-1.0 ; From Telefonica context=from-pstn signalling=pri_cpe accountcode=inbound echocancel=128 echocancelwhenbridged=yes echotraining=800 faxdetect=no group=0 channel = 1-15,17-31 ; from PABX signalling=pri_net context=from-pabx accountcode=outbound echocancel=128 echocancelwhenbridged=yes echotraining=800 faxdetect=no group=1 channel = 32-46,48-62 # lsmod Module Size Used by wct4xxp63168 62 zaptel207364 127 wct4xxp radeon125637 2 md5 4033 1 ipv6 234881 14 autofs423237 0 tun 9153 1 sunrpc159269 1 crc_ccitt 2113 1 zaptel microcode 6881 0 dm_mirror 27825 0 dm_mod 56661 1 dm_mirror hw_random 5845 0 e1000 93101 0 floppy 58481 0 ext3 116809 2 jbd71385 1 ext3 ata_piix9413 3 libata 44957 1 ata_piix sd_mod 17217 4 scsi_mod 121293 2 libata,sd_mod #cat /proc/interrupts = CPU0 0: 75799765 XT-PIC timer 1: 10 XT-PIC i8042 2: 0 XT-PIC cascade 3:6675655 XT-PIC eth0, [EMAIL PROTECTED]::06:05.0 5: 76837 XT-PIC libata 8: 1 XT-PIC rtc 10: 75733239 XT-PIC wct2xxp 12: 58 XT-PIC i8042 14: 681710 XT-PIC ide0 NMI: 0 ERR: 0 #cat /proc/zaptel/1 = Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 1 TE2/0/1/1 Clear (In use) 2 TE2/0/1/2 Clear (In use) 3 TE2/0/1/3 Clear (In use) 4 TE2/0/1/4 Clear (In use) 5 TE2/0/1/5 Clear (In use) 6 TE2/0/1/6 Clear (In use) 7 TE2/0/1/7 Clear (In use) 8 TE2/0/1/8 Clear (In use) 9 TE2/0/1/9 Clear (In use) 10 TE2/0/1/10 Clear (In use) 11 TE2/0/1/11 Clear (In use) 12 TE2/0/1/12 Clear (In use) 13 TE2/0/1/13 Clear (In use) 14 TE2/0/1/14 Clear (In use) 15 TE2/0/1/15 Clear (In use) 16 TE2/0/1/16 HDLCFCS (In use) 17 TE2/0/1/17 Clear (In use) 18 TE2/0/1/18 Clear (In use) 19 TE2/0/1/19 Clear (In use) 20 TE2/0/1/20 Clear (In use) 21 TE2/0/1/21 Clear (In use) 22 TE2/0/1/22 Clear (In use) 23 TE2/0/1/23 Clear (In use) 24 TE2/0/1/24 Clear (In use) 25 TE2/0/1/25 Clear (In use) 26 TE2/0/1/26 Clear (In use) 27 TE2/0/1/27 Clear (In use) 28 TE2/0/1/28 Clear (In use) 29 TE2/0/1/29 Clear (In use) 30 TE2/0/1/30 Clear (In use) 31 TE2/0/1/31 Clear (In use) # cat /proc/zaptel/2 = Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 32 TE2/0/2/1 Clear (In use) 33 TE2/0/2/2 Clear (In use) 34 TE2/0/2/3 Clear (In use) 35 TE2/0/2/4 Clear (In use) 36 TE2/0/2/5 Clear (In use) 37 TE2/0/2/6 Clear (In use) 38 TE2/0/2/7 Clear (In use) 39 TE2/0/2/8 Clear (In use) 40 TE2/0/2/9 Clear (In use) 41 TE2/0/2/10 Clear (In use) 42 TE2/0/2/11 Clear (In use) 43 TE2/0/2/12 Clear (In use) 44 TE2/0/2/13 Clear (In use) 45 TE2/0/2/14 Clear (In use) 46 TE2/0/2/15 Clear (In use) 47 TE2/0/2/16 HDLCFCS (In use) 48 TE2/0/2/17 Clear (In use) 49 TE2/0/2/18 Clear (In use) 50 TE2/0/2/19 Clear (In use) 51 TE2/0/2/20 Clear (In use) 52 TE2/0/2/21 Clear (In use) 53
Re: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet
A_ Navone, You cannot use a Y connector on a data (ethernet) connection, you must use a switch or and older hub to accomplish this. Regards, Humberto 2 SIP phones on Y data connector on 1 ethernet - will that cause problems ? thx in advance _ Don’t just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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 ___ --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] PAP2 and ringing issues
Hi Aaron, I tried the progressinband=no and it worked great. Thanks for the tip. Humberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: 01 November 2005 17:17 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PAP2 and ringing issues Hi, I currently have several PAP2-NA units configured to an Asterisk box, everything works fine except from the fact that after dialing a number I can hear ringing tones. When I connect to the same Asterisk box using XLite or EyeBeam I hear only one, any ideas on what may be wrong on the PAP units? Hi Humberto, We had this problem with calls being sent to a PRI. The two ringtones were due to both an RTP audio stream being generated from the PRI (this is the one we wanted) and also a SIP 180 ringing response being sent by the same Asterisk server. I'm not sure why both are getting sent, in 1.0.7 I'm pretty sure they weren't. The fix was simply to set progressinband=no in sip.conf on the Asterisk server with the PRI. The reason you only get the doble ring on one UA and not others seems to be entirely down to the UA. In our case the Linksys units act passed on both ringing indications where as Cisco IP Phones disregarded the SIP 180 and just passed on the RTP. Hth. Aaron ___ --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 ___ --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] PAP2 and ringing issues
Hi, I currently have several PAP2-NA units configured to an Asterisk box, everything works fine except from the fact that after dialing a number I can hear ringing tones. When I connect to the same Asterisk box using XLite or EyeBeam I hear only one, any ideas on what may be wrong on the PAP units? Thanks in advance, Humberto ___ --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] zaptel.conf config for CAS signalling
Even better, share the whole zaptel.conf Humberto would you please share line 213 with us? On 10/18/05, Matt Hess [EMAIL PROTECTED] wrote: I have a customer that needs to do cas signaling across a t1,esf span.. it looks like this can be done but I'm not sure how as the documentation is very light in regards to cas.. it would appear that I need to use sf signaling but I get an error saying: $ ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 213: Unknown keyword 'sf' I've also tried the format suggested in zaptel.conf channel# = (etc.) but I continue to fail.. I'd love a few pointers here.. ___ --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 ___ --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 ___ --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] Dial plan questions
Neil, When you use the Dial command you must specify the device to use for dialing, so you cannot use Dial(2201,20) you must use Dial(SIP/2201,20) which informs to use the the SIP device 2201. Regards, Humberto Aicardi I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten = 2201,1,Dial(sip/[EMAIL PROTECTED],20,) exten = 2201,2,Voicemail(u2201) exten = 2201,3,Hangup exten = 2201,102,voicemail(b2201) exten = 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as unavailable. This one does not work: exten = 2401,1,Dial(2201,20,) exten = 2401,2,Voicemail(u2201) exten = 2401,3,Hangup exten = 2401,102,voicemail(b2201) exten = 2401,103,hangup If I dial 2401 I get fast busy, what am I doing wrong? mozart*CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status 2202/2202 192.168.24.197 D 255.255.255.255 5060OK (17 ms) 2201/2201 (Unspecified)D 255.255.255.255 0 UNKNOWN pstn/pstn 192.168.24.197 D 255.255.255.255 5061OK (15 ms) ___ --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] Error when answering CAPI
Hi, I've a Fritz card which was working fine, recently I changed hardware and my nightmare started. Now when I call someone through the chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I always get hungup. Can someone please give some help? Here are the logs: *CLI -- CONNECT_IND ID=001 #0x LEN=0049 Controller/PLCI/NCCI= 0x101 CIPValue= 0x1 CalledPartyNumber = a11138121122 CallingPartyNumber = A831181114000 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=1138121122,CID=1181114000,CIP=0x1,CONTROLLER=0x1) -- creating pipe for PLCI=0x101 msn = * -- INFO_IND ID=001 #0x0001 LEN=0026 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = a11138121122 -- INFO_IND ID=001 #0x0001 LEN=0026 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = a11138121122 sent INFO_RESP (PLCI=0x101) sent CONNECT_RESP for PLCI = 0x101 -- INFO_IND ID=001 #0x0002 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 -- INFO_IND ID=001 #0x0002 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 sent INFO_RESP (PLCI=0x101) -- DISCONNECT_IND ID=001 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 == DISCONNECT_IND PLCI=0x101 REASON=0 sent DISCONNECT_RESP PLCI=0x101 -- CAPI Hangingup activehangingup -- removed pipe for PLCI = 0x101 capi.conf === ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ; mode: ptmp (point-to-multipoint) or ptp (point-to-point) isdnmode=ptp ; allow incoming calls to this list of MSNs, * == any incomingmsn=* ; capi controller number controller=1 ; dialout group group=1 ; enable/disable software dtmf detection, recommended for AVM cards softdtmf=1 ; accountcode to use in CDRs accountcode= ; context for incoming calls context=from-pstn ; _VERY_PRIMITIVE_ echo suppression ;echosquelch=1 ; EICON DIVA SERVER echo cancelation ;echocancel=yes ;echotail=64 ; call group ;callgroup=1 ; deflect incoming calls to 12345678 if all B channels are busy ;deflect=12345678 ; number of concurrent calls on this controller (2 makes sense for single BRI) devices = 2 ___ 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] Segmentation fault
Hi, I have just got to the office and now * is giving me the following error: -- Executing Macro(SIP/204-df20, dial|CAPI/111222:b130) in new stack -- Executing DBput(SIP/204-df20, RepeatDial/204=130) in new stack -- DBput: family=RepeatDial, key=204, value=130 -- Executing NoOp(SIP/204-df20, ) in new stack -- Executing Dial(SIP/204-df20, CAPI/111222:b130|30) in new stack Ouch ... error while writing audio data: : Broken pipe Segmentation fault I have tried everything, putting an old conf downloading the new head, but nothing. Friday night everything was working fine. I have even made a Shutdown/power-up procedure but still no success. My ISDN card is a Fritz PCI Card. Any help is surely appreciated since I am completely lost on this one. ; ; CAPI config ; ; [general] nationalprefix=11 internationalprefix=55 rxgain=0.0 txgain=0.0 [interfaces] msn=111222 incomingmsn=* controller=1 softdtmf=1 accountcode=capi context=from-capi echosquelch=1 ;echocancel=yes ;echotail=32 callgroup=1 ;deflect=12345678 devices=2 isdnmode=ptp ___ 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
RES: RES: [Asterisk-Users] chan_oh323 gatekeeper
But what happens if the H.323 device is on a dyamic IP? Do I need a gatekeeper? Thanks, Humberto -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Adi Linden Enviada em: Thursday, January 06, 2005 12:58 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: RES: [Asterisk-Users] chan_oh323 gatekeeper That's correct, just send FXO calls to the Asterisk box. Calls from H.323 sources will go into the context specified in the oh323.conf file. Adi On Wed, 5 Jan 2005, Humberto Aicardi wrote: You're right it works, but how about receiving calls, how can you register so the FXO gateways knows where to send the calls? Or I just setup the FXO gateway with the IP address of the * box? Humberto Aicardi ___ 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 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
RES: [Asterisk-Users] asterisk addson
I'm having exactly the same problem, I'm currently using * HEAD. If anyone can help please let us know what is the issue. Regards, Humberto -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de rizwan Enviada em: Thursday, January 06, 2005 4:34 AM Para: asterisk-users@lists.digium.com Assunto: [Asterisk-Users] asterisk addson Hello I am tying to install asterisk addson, but when i give make, it shows me following error messages: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Any idea, why this compatability issue arised? Thanks ___ 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 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] Asterisk and SER security doubts
Hi, I have configured * with a x100p and a E100 E1 card and everything is working fine, now I have setup a SER which the UA would connect, I will be using the * box as a E1 gateway and Voicemail. Anyway, I was alarmed after I tried the integration, when the SER forwards any call to the PSTN the * box wont check any credentials! Im a newbie so maybe this is the correct behavior for the SIP protocol, but when I try to connect directly to the * box it requires authentication, is there a way to setup * to require authentication? Can SER be registered in *? Any help would be much welcomed to better understand the integration of both softwares. Thanks in advance, Humberto ___ 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] chan_oh323 gatekeeper
Hi folks, Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi ___ 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
RES: [Asterisk-Users] chan_oh323 gatekeeper
You're right it works, but how about receiving calls, how can you register so the FXO gateways knows where to send the calls? Or I just setup the FXO gateway with the IP address of the * box? Humberto Aicardi -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Adi Linden Enviada em: Wednesday, January 05, 2005 9:28 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] chan_oh323 gatekeeper Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? I have this in my extensions.conf. The oh323.conf has gatekeeper disabled and nothing else specific to the 192.168.99.83. Works just fine to place calls to a Cisco fxo gateway. exten = s,3,Dial(OH323/[EMAIL PROTECTED]) ; H.323 Protocol Adi ___ 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 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] Echo problems
Hi to all, I recently installed a Fritz PCI ISDN BRI card in hope that the echo problems would go away, but unfortunately it didn't solve the problem when calling analog lines, when calling cell phones and PBX with a E1/T1 it works great. Can anyone suggest what would be a way to eliminate the echo problem using a BRI line. Would a BRI gateway solve the echo problem? Is there a PCI ISDN BRI card with echo cancellation? Please send some help. Thanks in advance, Humberto Aicardi ___ 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] ISDN HFC cards
Hi, Currently I am using a ISDN BRI PCI FRITZ card (works), would I get any benefits switching to a HFC card? Or it would be a better choice to switch to a ISDN with a DSP processor? Currently I have echo on my CAPI channel when calling analog lines, if call a cell phone, ISDN or a PRI PBX it doesn't show up any echo. So this indicates a far-end echo, how can this be minimized? I turned on the Squelch on the capi and it works but during a conversation the sometimes I tend to get small click and it distracts a little bit, even tough it is still better than the echo. If switching to HFC works better can someone point out where to buy them (online)? Regards, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Fax detection CAPI (doesn't work!)
Hi, Thanks for the reply, what I currently have is the chan_0.3.5 with the CVS HEAD and fax patch applied. I have made all combinations possible with softdtmf=0/1 and enabling/disabled FORCE_SOFTWARE_DTMF and nothing. During the loading process of the capi channel driver I get a CAPI[contrX] supports DTMF but still the extension for fax does not get executed, any toughts? Regards, Humberto Aicardi -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Carl Sempla Enviada em: Tuesday, December 14, 2004 8:15 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Fax detection CAPI (doesn't work!) On Tuesday, 14 December, 2004 22:17 : Humberto Aicardi [EMAIL PROTECTED] wrote: I'm currently using a ISDN-BRI with a Fritz ISDN card and the chan-capi. The problem is that the fax detection is not executed, Hi, The fax detection in chan_capi use the CAPI DTMF feature. So you need to set in /etc/asterisk/capi.conf the line softdtmf=0 Check if CFLAGS+=-DFORCE_SOFTWARE_DTMF in Makefile of chan_capi is commented (#). If you start asterisk with the -v option, you should see : CAPI[contrX] supports DTMF And obviously the card must report a DTMF when a fax tone is detected. It may be possible to alter the code and use the asterisk dsp code instead. Regards, -- Carl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax detection CAPI (doesn't work!)
Hi,
I'm currently using a ISDN-BRI with a Fritz ISDN card and the
chan-capi. The problem is that the fax detection is not executed, here is a
snippet of the extensions.conf:
[from-capi]
exten = h,1,Macro(record-cleanup)
exten = s,1,Wait(2)
exten = s,2,ResponseTimeout(15)
exten = s,3,Background(fn-intro)
exten = fax,1,capiAnswerFax(/tmp/${UNIQUEID}) ; --- never gets executed?!
Can someone provide me with further information if need to setup
anything else?
Best regards,
Humberto Aicardi
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip phone?
Hi, I currently have a * server with a IAXy adapter and a Voip phone. My doubt is: which is the best option? I personally find IAXy to be very effective, except from the fact that they don't support G729. The other option would be to use the TDM400P, which I have heard that it has some problems with echo, is this true? And finally to use a VOIP phone which look good and includes several extra features. Oops, I forgot there's still the gateway option, including ATA186, VoicePlanet, Mediatrix and so on. The problem is that they are expensive compared to prior options, except the VOIP phone. What I really need is a solution that works without the usual * echo problems. The major issue with IAXy is the price at US$99. I can buy for US$ 75 a Grandstream BT102. Can anyone share their experience with the above solutions? Thanks in advance, Humberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with CDR and the dst field!
Hi,
I currently have a context similar to this one:
[test]
exten = h,1,Macro(record-cleanup)
exten =
1000,1,Macro(record-enable,${EXTEN},${CALLERIDNUM})
exten = 1000,2,Dial
Everything works great, except that now I get a h on the CDR dst
field! I am using cdr_addons_mysql.so, is there any way to prevent this from
happening?
Regards,
Humberto Aicardi
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Question about e1/digium
Sergio, I currently live in São Paulo, and have done exactly that configuration. Where I have an E1 (PRI) with 15 lines and 100 (DDR) numbers. If you are using Embratel as a E1 provider then the DID they send is 4 bytes long. All you have to do is this: Exten = 0001,1,Dial(SIP/1000) ; All calls routed to 4000-0001 would be dial SIP extension 1000 Exten = 0010,1,Dial(SIP/1002SIP/1004) ; In this case all calls to 4000-0010 would ring two extensions Hope this helps! Humberto Aicardi [EMAIL PROTECTED] -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de SERGIO GUIMARAES FAULHABER Enviada em: Tuesday, December 07, 2004 9:36 AM Para: [EMAIL PROTECTED] Assunto: [Asterisk-Users] Question about e1/digium Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to direct for a telephone sip ? Thanks for attencion Sergio Faulhaber [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI Newbie
Hi everyone,
Hi, I am new to the ISDN world, today I received my new NT box from
CS Telecom but unfortunately it doesn't work. I have installed all the
drivers and when I issue a capiinfo I get:
Number of Controllers: 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02(49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
---
When I issue within * capi info I get:
Contr1: 2 B channels total, 2 B channels free.
But when I dial to the number nothing happens, when I dial using capi I get:
Dec 3 19:26:55 WARNING[2585]: chan_capi.c:653 capi_call: Destination
38126927 requres a real destination
-- Couldn't call 38126927
-- CAPI Hangingup
Finally I have a Fritz card, which I plugged into the NT S/To bus using a
plain Ethernet cable. Is there a way to know if the Fritz PCI card is
talking to the NT device?
Any help would be welcomed.
Thanks in advanced,
Humberto Aicardi
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
