Re: [asterisk-users] Does Asterisk-16.1.1 support "make freepbx"
On Sun, 30 Dec 2018 at 15:10, Joshua C. Colp wrote: > > On Sun, Dec 30, 2018, at 9:03 AM, james wrote: > > hello: > > Does asterisk-16.1.1 support freepbx by default? > > No version of Asterisk currently has any built in mechanism to install and > set up FreePBX. They operate as separate projects and the FreePBX install > instructions would need to be used to install it. > The article in the original post was published on December 28th when Spain celebrates their April's Fools. Greetings, Jorge -- Jorge Martínez López -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/2.0 401 Unauthorized when calling from one SIP extension to another
Hello everyone, A few days ago I had a problem with a couple of extensions. I have about 12 Aastra 6731i phones, 6 are at our main office and 6 more on remote branches. We use VPN to communicate to our branches so there's no NAT involved any where. The problem I had was that I couldn't call from two extensions located at two branch offices. But I could call to them just fine. On any call placed from those phones I got the following error: SIP/2.0 401 Unauthorized This is the console output of a call placed from one of those phones: --- SIP read from UDP:192.168.96.141:5060 --- INVITE sip:85004@192.168.10.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.96.141:5060 ;branch=z9hG4bKd4511cd84a4f22669.bbb2635a0516602e8 Max-Forwards: 70 From: sip:85014@192.168.10.227:5060;tag=5dde10fb77 To: 85004 sip:85004@192.168.10.227:5060 Call-ID: 169216acc663493c CSeq: 28267 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: sip:85014@192.168.96.141:5060 ;transport=udp;+sip.instance=urn:uuid:--1000-8000-00085D2B85C3 Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 6731i/2.6.0.1007 Content-Type: application/sdp Content-Length: 698 v=0 o=MxSIP 0 0 IN IP4 192.168.96.141 s=SIP Call c=IN IP4 192.168.96.141 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 4 4 98 97 115 96 9 108 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:108 G7221/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv - --- (14 headers 29 lines) --- Sending to 192.168.96.141:5060 (no NAT) Sending to 192.168.96.141:5060 (no NAT) Using INVITE request as basis request - 169216acc663493c Found peer '85014' for '85014' from 192.168.96.141:5060 --- Reliably Transmitting (NAT) to 192.168.96.141:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.96.141:5060 ;branch=z9hG4bKd4511cd84a4f22669.bbb2635a0516602e8;received=192.168.96.141;rport=5060 From: sip:85014@192.168.10.227:5060;tag=5dde10fb77 To: 85004 sip:85004@192.168.10.227:5060;tag=as52309181 Call-ID: 169216acc663493c CSeq: 28267 INVITE Server: Asterisk PBX 11.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=03eab1fd Content-Length: 0 Scheduling destruction of SIP dialog '169216acc663493c' in 32000 ms (Method: INVITE) --- SIP read from UDP:192.168.96.141:5060 --- ACK sip:85004@192.168.10.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.96.141:5060 ;branch=z9hG4bKd4511cd84a4f22669.bbb2635a0516602e8 Max-Forwards: 70 From: sip:85014@192.168.10.227:5060;tag=5dde10fb77 To: 85004 sip:85004@192.168.10.227:5060;tag=as52309181 Call-ID: 169216acc663493c CSeq: 28267 ACK User-Agent: Aastra 6731i/2.6.0.1007 Content-Length: 0 And that just keep repeating and repeating but the call never actually takes place. The contents of my sip.conf file: [general] context=unauthenticated allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no shrinkcallerid=no [office-phone](!) type=peer context=LocalSets host=dynamic nat=force_rport,comedia dtmfmode=auto disallow=all allow=g729 [85004](office-phone) defaultuser=85004 secret=securepass callerid=Phone 4 85004 [85014](office-phone) defaultuser=85014 secret=securepass callerid=Phone 14 85014 host=192.168.96.141 transport=udp,tcp Originally I had not have the defaultuser option on any of the extensions, nor the host and transport on the [85014] one, but the problem was the same with or without those options. Note that I'm including only two extensions to simplify things up and that the extension with the problem is 85014. Also, I said there's no NAT involved here but I'm using the option nat=force_rport,comedia as suggested by Asterisk The Definitive Guide 4th edition. I've also switched that option to nat=no and the result was been the same. My dialplan is also really simple. extensions.conf file: [LocalSets] exten = 85004,1,Dial(SIP/85004) exten = 85014,1,NoOp() same =
[asterisk-users] Retransmission
Hello, When I try to call outside, receive this message: [2013-02-14 10:11:28] WARNING[7440]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 753559dd5cbd4aef5f42ef3a414892b9@X.X.X.X:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ... I want help to fix it regards -- Jorge Quitério IT Specialist unix.co.ao Linux User: #533142 jquiteri...@gmail.com +244 927 161 667 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to control just one phone within current CCM?
Hi, I have used basic Asterisk as a PBX controlling few extensions. My question is simple, at work there is an existing Call Manager/PBX and all which manages all the extensions for SCCP VOIP phones. Can Asterisk be used to manage just 1 VOIP phone and still can make internal calls to the other extensions? Thanks, Jorge -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] - SIP retransmission problem
Hi Paolo, I had yesterday a similar problem and it was caused by a misconfigured IP address in extensions.conf that I forgot to update after changing some IP addresses in my network. Check the network connectivity between you Asterisk host and 1000. Double check that the IP address is correct. Use tcpdump to see what's going on the wires. Good luck! -- Jorge Martínez López jorg...@gmail.com http://www.jorgeml.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi+Redfone+Channel Bank+EM
Thank you again Mitul. Ok, the we ill use EM. Regards Jorge Mendoza - Original Message - From: Mitul Limbani mi...@enterux.in To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 25 July, 2012 11:35:08 PM Subject: Re: [asterisk-users] Dahdi+Redfone+Channel Bank+EM Same for E1 as well unless your operator is giving mfcr2 on cas. Mitul On Jul 26, 2012 9:17 AM, Jorge Mendoza jmendo...@tcc.com.pe wrote: Thank you Mitul for your answer. Yes, we have tested em before and it works. But I don't know why. That is what I don't understand. You said that EM signalling does not have separate signalling channel, that is true for the T1 but not for E1. My understanding is that E1 CAS pass the EM information in the bits abcd of channel 16, the signalling channel. Regards -- Jorge Mendoza From: Mitul Limbani mi...@enterux.in To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 25 July, 2012 8:15:25 PM Subject: Re: [asterisk-users] Dahdi+Redfone+Channel Bank+EM EM signalling do not have seperate signalling channel. Configure as em=1-31 Mitul On Jul 26, 2012 6:40 AM, Jorge Mendoza jmendo...@tcc.com.pe wrote: blockquote Hi, We are trying to connect an Asterisk server with a Channel Bank with EM interfaces using a RedFone TDMoE device. The CB have a E1 CAS interface. OS: Ubuntu Server 11.10 64 bits dahdi: dahdi-linux-complete-2.6.1+2.6.1 Redfone configuration: /etc/redfone.conf [span1] framing=cas encoding=hdb3 System configuration: /etc/dahdi/system.conf dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1 dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0 eme1=1-15,17-31 dchan=16 alaw=1-31 loadzone = fr defaultzone = fr Error message: # dahdi_cfg -v Changing signalling on channel 1 from Unused to E M E1 Changing law on channel 1 from Mu-law to A-law DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: eme1 signaling is being used on a T1 line (use em) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span = I don't understand the last possible cause mentioned: Signaling is being assigned to channel 16 of an E1 CAS span, because the dchan is channel 16. Where is the error? Thank you. -- Jorge Mendoza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users /blockquote -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi+Redfone+Channel Bank+EM
Hi, We are trying to connect an Asterisk server with a Channel Bank with EM interfaces using a RedFone TDMoE device. The CB have a E1 CAS interface. OS: Ubuntu Server 11.10 64 bits dahdi: dahdi-linux-complete-2.6.1+2.6.1 Redfone configuration: /etc/redfone.conf [span1] framing=cas encoding=hdb3 System configuration: /etc/dahdi/system.conf dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1 dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0 eme1=1-15,17-31 dchan=16 alaw=1-31 loadzone = fr defaultzone = fr Error message: # dahdi_cfg -v Changing signalling on channel 1 from Unused to E M E1 Changing law on channel 1 from Mu-law to A-law DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: eme1 signaling is being used on a T1 line (use em) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span = I don't understand the last possible cause mentioned: Signaling is being assigned to channel 16 of an E1 CAS span, because the dchan is channel 16. Where is the error? Thank you. -- Jorge Mendoza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi+Redfone+Channel Bank+EM
Thank you Mitul for your answer. Yes, we have tested em before and it works. But I don't know why. That is what I don't understand. You said that EM signalling does not have separate signalling channel, that is true for the T1 but not for E1. My understanding is that E1 CAS pass the EM information in the bits abcd of channel 16, the signalling channel. Regards -- Jorge Mendoza - Original Message - From: Mitul Limbani mi...@enterux.in To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 25 July, 2012 8:15:25 PM Subject: Re: [asterisk-users] Dahdi+Redfone+Channel Bank+EM EM signalling do not have seperate signalling channel. Configure as em=1-31 Mitul On Jul 26, 2012 6:40 AM, Jorge Mendoza jmendo...@tcc.com.pe wrote: Hi, We are trying to connect an Asterisk server with a Channel Bank with EM interfaces using a RedFone TDMoE device. The CB have a E1 CAS interface. OS: Ubuntu Server 11.10 64 bits dahdi: dahdi-linux-complete-2.6.1+2.6.1 Redfone configuration: /etc/redfone.conf [span1] framing=cas encoding=hdb3 System configuration: /etc/dahdi/system.conf dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1 dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0 eme1=1-15,17-31 dchan=16 alaw=1-31 loadzone = fr defaultzone = fr Error message: # dahdi_cfg -v Changing signalling on channel 1 from Unused to E M E1 Changing law on channel 1 from Mu-law to A-law DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: eme1 signaling is being used on a T1 line (use em) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span = I don't understand the last possible cause mentioned: Signaling is being assigned to channel 16 of an E1 CAS span, because the dchan is channel 16. Where is the error? Thank you. -- Jorge Mendoza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering machine answers after pickup a phone.
Hi again, thanks for your answer, but it didn't solve the problem. That Dial command returns inmediately, so I don't even have the delay. I'll try to explain myself better. The PBX has only one FXO card, connected to the PSTN. There is no other phones connected to the PBX nor SIP extensions. There are analog phones connected to the same PSTN. What I try to do is that, when there is an incoming call from the ouside, if someone answers on a phone, then the PBX won't answer. Thanks. O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu: Hi, your concept using Wait() won't work here. Try it like this: [incoming] exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s exten = s,n,BackGround(wellcome-message) exten = s,n,Voicemail(1234) exten = #,1,Hangup() So, of you answer the call within 30s, you'll get the call on your phone. After 30s, the Voicemail will answer the phone. regards, Ruben Am 04.08.2011 21:39, schrieb Jorge Barreiro: Hello, I'm configuring an Asterisk PBX to use as an answering machine. I have a FXO card connected to the line, and other analog telephones connected to the same line. The PBX answers and redirects you to the voicemail after a delay. The problem is that even if I pickup any analog phone in the line before the PBX does, it answers after the delay anyway. And I couldn't find how to prevent this, or even if this is supposed to happen. My FXO card is a cheap X100P (source of problems, I know), and I'm using the Asterisk version included in Debian Squeeze (1.6.2.9). My dial plan looks like this: [incoming] exten = s,1,Wait(8) exten = s,2,Answer exten = s,3,BackGround(wellcome-message) exten = s,4,Voicemail(1234) exten = #,1,Hangup I don't know if this is related, but I'm in Spain and I had to add: hanguponpolarityswitch=yes to the chan_dahdi.conf file so that asterisk detects the remote hangup. I also added: answeronpolarityswitch=yes but this didn't help. It seems to be used just to detect the answer when you are calling, not when receiving a call. I'd appreciate any help you could provide. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering machine answers after pickup a phone.
Hi, thanks for your time! O Venres, 5 de Agosto de 2011 12:35:05 escribiches: Completely normal operation. You need to read and understand more basic telephony and analog lines to understand why that won't work. I definitely have a lot to learn yet. Asterisk needs to be in control, and once someone answers a phone not under Asterisk control, or the call is abandoned there is little you can do. What I pretend is that asterisk detects that it's not under control and gets out of the way. The same way it detects a remote hangup and stops the dialplan, it could detect that someone else answered (the line is not ringing anymore) and discard it the same way it does when the remote part hangup. I've read comments in forums and tutorials that seem to imply that this happens, but I couldn't find any confirmation (and indeed, it's not happening to me). If you confirm me that this is the normal behavior, then I at least I know my solution is in the dialplan and not a card/line/driver problem. Sounds like a task for a simple answering machine from Wal-Mart All you other phones should be connected to FXS ports, or you need to be smarter in your dialplan. Once you answer, Asterisk is behaving normally Yes, it's a really simple task, but this should be just a starting point. The plan is to start migrating services to the PBX little by little, and the voicemail looked like the easier thing to start. I wanted to maintain the current analog phones until I feel confident with the asterisk configuration. Maybe it wasn't such a great idea, and I should start by moving the phones to FXS ports in the PBX. John Novack Jorge Barreiro wrote: Hi again, thanks for your answer, but it didn't solve the problem. That Dial command returns inmediately, so I don't even have the delay. I'll try to explain myself better. The PBX has only one FXO card, connected to the PSTN. There is no other phones connected to the PBX nor SIP extensions. There are analog phones connected to the same PSTN. What I try to do is that, when there is an incoming call from the ouside, if someone answers on a phone, then the PBX won't answer. Thanks. O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu: Hi, your concept using Wait() won't work here. Try it like this: [incoming] exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s exten = s,n,BackGround(wellcome-message) exten = s,n,Voicemail(1234) exten = #,1,Hangup() So, of you answer the call within 30s, you'll get the call on your phone. After 30s, the Voicemail will answer the phone. regards, Ruben Am 04.08.2011 21:39, schrieb Jorge Barreiro: Hello, I'm configuring an Asterisk PBX to use as an answering machine. I have a FXO card connected to the line, and other analog telephones connected to the same line. The PBX answers and redirects you to the voicemail after a delay. The problem is that even if I pickup any analog phone in the line before the PBX does, it answers after the delay anyway. And I couldn't find how to prevent this, or even if this is supposed to happen. My FXO card is a cheap X100P (source of problems, I know), and I'm using the Asterisk version included in Debian Squeeze (1.6.2.9). My dial plan looks like this: [incoming] exten = s,1,Wait(8) exten = s,2,Answer exten = s,3,BackGround(wellcome-message) exten = s,4,Voicemail(1234) exten = #,1,Hangup I don't know if this is related, but I'm in Spain and I had to add: hanguponpolarityswitch=yes to the chan_dahdi.conf file so that asterisk detects the remote hangup. I also added: answeronpolarityswitch=yes but this didn't help. It seems to be used just to detect the answer when you are calling, not when receiving a call. I'd appreciate any help you could provide. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] Answering machine answers after pickup a phone.
O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu: On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote: Hi, thanks for your time! O Venres, 5 de Agosto de 2011 12:35:05 escribiches: Completely normal operation. You need to read and understand more basic telephony and analog lines to understand why that won't work. I definitely have a lot to learn yet. Asterisk needs to be in control, and once someone answers a phone not under Asterisk control, or the call is abandoned there is little you can do. What I pretend is that asterisk detects that it's not under control and gets out of the way. The same way it detects a remote hangup and stops the dialplan, it could detect that someone else answered (the line is not ringing anymore) and discard it the same way it does when the remote part hangup. I've read comments in forums and tutorials that seem to imply that this happens, but I couldn't find any confirmation (and indeed, it's not happening to me). When I first installed Asterisk in my home I used it in the way that you described: as a glorified answering machine to email to me any voice mail. I think what you want is the WaitForRing()[1] dial plan application. This function will wait x number of seconds, then look for *another* ring to come in. If someone answered the phone before the timeout to that function Asterisk would stop processing the dial plan. [1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing I ran into a couple of issues with WaitForRing(). The first being if someone answered the phone and then quickly hung up *and* a new phone call came in within the timeout period, Asterisk wouldn't know that the line was ringing due to a new call. The second problem was I never got the dial tone detection working so that if I tried to *place* a call from Asterisk while someone was on the house line I would aggravate my wife. Since coming to work for Digium I've seen in the data sheets for the FXO interfaces that there is a capability to detect when a parallel device on a line goes off hook. This would allow Asterisk to have a better sense of the state of the line (like it currently can detect when a port is unplugged and there is not battery by generating a red alarm.) but I haven't looked into getting that information off the hardware and up into Asterisk. Hope this helps, Shaun That application looks like a good solution. I can't test it until Monday, but I'll try it and let you know. The drawbacks you mention doesn't seem too inconvenient in my case. Anyway, I started with this cause I thought it was an easy first step, if it gets so complicated I think I'll go forward and put all phones under the control of the PBX. Thank you everybody for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering machine answers after pickup a phone.
O Venres, 5 de Agosto de 2011 21:20:37 Don Kelly escribiu: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Barreiro Sent: Friday, August 05, 2011 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Answering machine answers after pickup a phone. O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu: On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote: Hi, thanks for your time! O Venres, 5 de Agosto de 2011 12:35:05 escribiches: Completely normal operation. You need to read and understand more basic telephony and analog lines to understand why that won't work. I definitely have a lot to learn yet. Asterisk needs to be in control, and once someone answers a phone not under Asterisk control, or the call is abandoned there is little you can do. What I pretend is that asterisk detects that it's not under control and gets out of the way. The same way it detects a remote hangup and stops the dialplan, it could detect that someone else answered (the line is not ringing anymore) and discard it the same way it does when the remote part hangup. I've read comments in forums and tutorials that seem to imply that this happens, but I couldn't find any confirmation (and indeed, it's not happening to me). When I first installed Asterisk in my home I used it in the way that you described: as a glorified answering machine to email to me any voice mail. I think what you want is the WaitForRing()[1] dial plan application. This function will wait x number of seconds, then look for *another* ring to come in. If someone answered the phone before the timeout to that function Asterisk would stop processing the dial plan. [1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing I ran into a couple of issues with WaitForRing(). The first being if someone answered the phone and then quickly hung up *and* a new phone call came in within the timeout period, Asterisk wouldn't know that the line was ringing due to a new call. The second problem was I never got the dial tone detection working so that if I tried to *place* a call from Asterisk while someone was on the house line I would aggravate my wife. Since coming to work for Digium I've seen in the data sheets for the FXO interfaces that there is a capability to detect when a parallel device on a line goes off hook. This would allow Asterisk to have a better sense of the state of the line (like it currently can detect when a port is unplugged and there is not battery by generating a red alarm.) but I haven't looked into getting that information off the hardware and up into Asterisk. Hope this helps, Shaun That application looks like a good solution. I can't test it until Monday, but I'll try it and let you know. The drawbacks you mention doesn't seem too inconvenient in my case. Anyway, I started with this cause I thought it was an easy first step, if it gets so complicated I think I'll go forward and put all phones under the control of the PBX. Thank you everybody for your help. I don't think this is a solution to the problem you described. No matter how long Asterisk 'waits for ring,' if the call has already been answered when Asterisk picks up, things won't work out well. The idea is that asterisk doesn't pick up if doesn't find the ring. The solution I described earlier, adding a simple exclusion device, will preclude Asterisk 'stepping on' a call in progress. This is the approach that Shaun suggests: ...a capability to detect when a parallel device on a line goes off hook. As it has not been implemented in Asterisk, it can be handled by an inexpensive device. This will enable you to do as you planned--test your implementation step-by-step, starting with the answering machine. Yes, that exclusion device would be more of a solution instead of just a workaround. But I'm finding it hard to find where to buy one in Spain (I've just started to look for them, anyway). Thanks. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users
[asterisk-users] Answering machine answers after pickup a phone.
Hello, I'm configuring an Asterisk PBX to use as an answering machine. I have a FXO card connected to the line, and other analog telephones connected to the same line. The PBX answers and redirects you to the voicemail after a delay. The problem is that even if I pickup any analog phone in the line before the PBX does, it answers after the delay anyway. And I couldn't find how to prevent this, or even if this is supposed to happen. My FXO card is a cheap X100P (source of problems, I know), and I'm using the Asterisk version included in Debian Squeeze (1.6.2.9). My dial plan looks like this: [incoming] exten = s,1,Wait(8) exten = s,2,Answer exten = s,3,BackGround(wellcome-message) exten = s,4,Voicemail(1234) exten = #,1,Hangup I don't know if this is related, but I'm in Spain and I had to add: hanguponpolarityswitch=yes to the chan_dahdi.conf file so that asterisk detects the remote hangup. I also added: answeronpolarityswitch=yes but this didn't help. It seems to be used just to detect the answer when you are calling, not when receiving a call. I'd appreciate any help you could provide. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with System() application
Are you able to execute: log.sh through the asterisk user?
On Wed, 20 Jul 2011 14:53:53 -0300, Antonio Modesto
mode...@isimples.com.br wrote:
Good afternoon,
I am trying to use the System() application but it is always
returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to run
this command:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt}
${exten});
This is the content of the /var/spool/asterisk/calllog/log.sh:
#!/bin/sh
#
#
TIME=$(date +%d-%m-%Y-%HH-%MM)
SOURCE=$1
DST=$2
echo $TIME - $SOURCE - $DST teste.log
I tried to insert some info direct into the file using echo but i've got
the same error.
Is there some secret to use this? haha
--
Jorge Gutiérrez
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[asterisk-users] Playback noanswer SIP
Hi, I would to send a message to an incoming call with no answer. My Asterisk server receive the incoming call by a BRI/SIP gateway (Could be a PRI, for instance). I do the command playback with option noanswer, Asterisk send 183 followed by RTP and finish with 603. But the BRI gateway do not allow to pass the RTP without a 200 OK. The question is: are there a SIP command to indicate the gateway to allow pass the RTP without the 200? This is an usual case when the Service Provider play a message like I'm sorry, you have dialed a wrong number So, I assume that the SIP protocol have foreseen the commands to implement this feature, I hope. Thank You -- Jorge Mendoza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver
Redfone uses and improved, in house developed TDMoE driver, officially supported by same Redfone. Redfone´s support site maintains tdmoe driver updated and certified to operate in every zaptel and dahdi versions. Txs Jorge Churio Redfone Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime LDAP Queues crashes
On Friday 08 January 2010 01:38:42 Gavin Henry wrote: What are the LDAP searches like? after updating and applying this patch: http://issues.asterisk.org/view.php?id=13573 doesn't crash and the queries i get are ok: conn=0 op=67 SRCH base=dc=nodomain scope=2 deref=0 filter=((objectClass=AsteriskQueue)(AstQueueName=barbaros)) = bdb_equality_candidates: (AstQueueName) not indexed conn=0 op=67 ENTRY dn=cn=barbaros,ou=queues,dc=nodomain conn=0 op=67 SEARCH RESULT tag=101 err=0 nentries=1 text= conn=0 op=68 SRCH base=dc=nodomain scope=2 deref=0 filter=((objectClass=AsteriskQueueMember)(AstQueueInterface=*) (AstQueueMemberof=barbaros)) = bdb_equality_candidates: (AstQueueMemberof) not indexed conn=0 op=68 ENTRY dn=uid=1234,ou=users,dc=nodomain conn=0 op=68 ENTRY dn=uid=demo,ou=users,dc=nodomain conn=0 op=68 SEARCH RESULT tag=101 err=0 nentries=2 text= but the queue is shown as empty: -- Executing [...@users:1] Queue(SIP/jsalamero-0001, barbaros) in new stack [Jan 12 16:32:37] WARNING[4238]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:32:37] WARNING[4238]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo -- Started music on hold, class 'default', on channel 'SIP/jsalamero-0001' voip*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Realtime 1234/1234 87.222.XXX.XXX D N 5060 OK (91 ms) Cached RT jsalamero/jsalamero87.222.XXX.XXX D N 1024 OK (86 ms) Cached RT /94.23.xxx.xxx5060 Unmonitored 3 sip peers [Monitored: 2 online, 0 offline Unmonitored: 1 online, 0 offline] voip*CLI queue show barbaros barbaros has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s No Members Callers: 1. SIP/jsalamero-0001 (wait: 0:44, prio: 0) [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo after adding by hand the users 1234 and demo to the queue, it works: queue add member SIP/demo to barbaros queue add member SIP/1234 to barbaros voip*CLI queue show barbaros barbaros has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:2, SL:0.0% within 0s Members: SIP/demo (dynamic) (Not in use) has taken no calls yet SIP/1234 (dynamic) (Not in use) has taken no calls yet No Callers voip*CLI [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo -- Executing [...@users:1] Queue(SIP/jsalamero-0005, barbaros) in new stack [Jan 12 16:42:51] WARNING[4754]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:42:51] WARNING[4754]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo -- Started music on hold, class 'default', on channel 'SIP/jsalamero-0005' -- SIP/demo-0007 is ringing -- SIP/1234-0006 is ringing -- Stopped music on hold on SIP/jsalamero-0005 == Spawn
[asterisk-users] Realtime LDAP Queues crashes
Hi all,
I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other
attributes needed for a working LDAP backend (I'll open a bug to include these
changes on svn).
SIP users and dialplan are perfectly working, but when I call a queue the
whole Asterisk (1.6.2.0) crashes:
on extconfig:
[settings]
sipusers = ldap,dc=nodomain,sip
sippeers = ldap,dc=nodomain,sip
extensions = ldap,dc=nodomain,extensions
voicemail = ldap,dc=nodomain,voicemail
queue_members = ldap,dc=nodomain,queue_member
queues = ldap,dc=nodomain,queue
on res_ldap.conf: see [1]
for the Queues on LDAP I have:
ou=Queues,dc=nodomain
ou: Queues
objectClass: top
objectClass: organizationalUnit
cn=foobar,ou=Queues,dc=nodomain
objectClass: applicationProcess
objectClass: AsteriskQueue
AstQueueName: foobar
AstQueueContext: default
AstQueueTimeout: 180
cn: foobar
the dialplan (on extensions.conf, the same if it's on LDAP):
[frontdesk]
exten = 78,1,Answer
exten = 78,n,Queue(foobar)
exten = 78,n,Hangup
[default]
include = common
include = frontdesk
switch = Realtime
and the user on LDAP:
uid=foo,ou=Users,dc=nodomain
cn: foo foo
uid: foo
sn: foo
uidNumber: 2002
gidNumber: 1901
homeDirectory: /nonexistent
userPassword: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
eboxSha1Password: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
eboxMd5Password: {MD5}rL0Y20zC+Fzt72VPzMSk2A==
eboxLmPassword: 5BFAFBEBFB6A0942AAD3B435B51404EE
eboxNtPassword: AC8E657F83DF82BEEA5D43BDAF7800CC
eboxDigestPassword: {MD5}x0Z+Prb70OIF3iARsuJ3Xg==
eboxRealmPassword: {MD5}c7467e3eb6fbd0e205de2011b2e2775e
givenName: foo
description: foo
AstAccountType: friend
AstAccountContext: users
AstAccountCallerID: 1001
AstAccountMailbox: 1001
AstAccountHost: dynamic
AstAccountNAT: yes
AstAccountQualify: yes
AstAccountCanReinvite: no
AstAccountDTMFMode: rfc2833
AstAccountInsecure: port
AstAccountLastQualifyMilliseconds: 0
AstAccountIPAddress: 0.0.0.0
AstAccountPort: 0
AstAccountExpirationTimestamp: 0
AstAccountRegistrationServer: 0
AstAccountUserAgent: 0
AstAccountFullContact: sip:0.0.0.0
AstContext: users
AstVoicemailMailbox: 1001
AstVoicemailPassword: 1001
AstVoicemailEmail: u...@domain
AstVoicemailAttach: yes
AstVoicemailDelete: no
AstQueueMembername: foobar
AstQueueMemberof: foobar
objectClass: AsteriskQueueMember
objectClass: AsteriskSIPUser
objectClass: AsteriskVoiceMail
objectClass: inetOrgPerson
objectClass: passwordHolder
objectClass: posixAccount
AstQueueInterface: SIP/1001
when i call the queue extension, on slapd I can see how Asterisk fetches the
AsteriskQueue objectClass, and then fetches the foo user, but then crashes
like this:
-- Executing [...@users:1] Answer(SIP/demo-, ) in new stack
-- Executing [...@users:2] Queue(SIP/demo-, foobar) in new
stack
[Jan 5 13:26:28] WARNING[6195]: app_queue.c:1134 create_queue_member: No
location at interface ''
[1]6124 segmentation fault (core dumped) asterisk -
vvc
*CLI queue show foobar
[1]6356 segmentation fault (core dumped) asterisk -
vvc
*CLI queue add member SIP/foo to foobar
[1]6394 segmentation fault (core dumped) asterisk -
vvc
any clue on what's wrong ? how could i debug this ? maybe there is some
attribute missing ? or the LDAP schema is wrong ? anyone with a working setup
like this ?
thanks in advance !
[0] http://people.ebox-platform.com/~bencer/asterisk.ldif
[1] http://people.ebox-platform.com/~bencer/res_ldap.conf.mas
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Re: [asterisk-users] No tone, one way communcation.
Once the card was configured correctly, have you set on the GUI the correct port to your zap extension? On Mon, 26 Oct 2009 05:02:18 -0700 (PDT), PATRICK KANGETHE patricemb...@yahoo.com wrote: 1. When i connected my analog phone to fxs card, i cannot get dial tone what could be the problem? I am using elastix 1.5.2 based on centos 5.2 Final. 2. On my 2 sip softphones using x-lite linux versions, i get one way audio how do i solve this?. This problem is also present when i use a windows version on one end and linux version on other end. Any help will be highly appreciated. -- Atentamente, Jorge Gutiérrez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Hardphones.
Yes I have used ATCOM-530 as an iax2 extension, without any trouble On Thu, 22 Oct 2009 17:33:13 +0100, Albert Culleton a...@icmunicomp.ie wrote: Hi there, Has anyone Used ATCOM IAX Hard phones with any success? Or Has anyone found any good IAX ATA that you could recommend. Thanks Albert This e-mail, as well as any other mode of correspondence, and any files transmitted with it are intended for, and should only be read by, the intended addressee. Its contents are confidential and if you are not the intended addressee, please notify the sender immediately and delete all records of the message from your computer. Any reproduction, dissemination, copying, disclosure, modification, distribution and/or publication of this message without the prior written consent of the sender are strictly prohibited. If you have received this message in error, please immediately notify the sender and delete the mail. This disclaimer will be modified without notice from time to time as new developments arises. Thank you. -- Atentamente, Jorge Gutiérrez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy on asterisk 1.6
Thanks very much, it worked as I needed :)
On Wed, 14 Oct 2009 17:14:53 +0530, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
hey In 1.6 version actually not wrote any code for option 'o'
you need to add following line into file
Index: apps/app_chanspy.c
===
--- apps/app_chanspy.c(revision 215998)
+++ apps/app_chanspy.c(working copy)
@@ -427,7 +427,12 @@
return -1;
}
- f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+ if (ast_test_flag(chan, OPTION_READONLY)) {
+ /* Option 'o' was set, so don't mix channel audio */
+ f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_READ, AST_FORMAT_SLINEAR);
+ } else {
+ f = ast_audiohook_read_frame(csth-spy_audiohook, samples,
AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+ }
ast_audiohook_unlock(csth-spy_audiohook);
regards
Dhaval
2009/10/14 Jorge Gutiérrez jgutier...@palosanto.com
I have read about that on asterisk 1.6, there will be a parameter o
(Only
listen to audio coming from this channel), I have tried, but I still get
inbound and outbound audio from the spied channel.
Has anyone used this feature? Is it working? Is there any work-around?
I will like to only spy the outbound audio from a channel, I dont want
to
hear the incomming audio of that channel.
I have used the following context:
[Conf]
exten = s,1,Answer
exten = s,2,Background(custom/menu_test)
exten = s,3,ChanSpy(,qoX)
exten = 1,1,Goto(Conf,s,2)
exten = 2,1,AGI(conf.php,${CALLERID(num)},${SPY_CHANNEL})
exten = 2,n,Goto(s,3)
exten = s,n,Goto(test2,s,1)
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Jorge Gutiérrez
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Re: [asterisk-users] PSTN to SIP line ratio
Shahnawaz Mir wrote: Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. Regards Smir _ You need to undertand traffic. See for instance: http://www.wirelesscommunication.nl/reference/chaptr04/erlang/erlang.htm Regards Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 IPs for an Asterisk server.
Yes it is possible, the only thing that you need to do is to configure correctly your network routes, if your ip devices are on the same net of your elastix you wont need to do any route configuration. Just leave the default gateway for your wan provider, it should work without any trouble On Thu, 15 Oct 2009 21:58:47 +0200, Guillaume Yziquel guillaume.yziq...@citycable.ch wrote: Hello. I've been setting up an Asterisk server, and I am now supposed to move it to a different network than the one it was set on. I'd like to give the server 2 IP address: -1- The first IP address is the IP it will have on the LAN, meaning that softphones will register to the Asterisk server using this 1st IP. -2- The second IP is the one that it will use to connect to the remote VoIP provider, which is using another network range than the LAN where I have my softphones. The default gateway would be the one of this second network address range. No NAT involved anywhere in this setup. Is it possible to do such a thing with Asterisk? Does it need really special tweaking of Asterisk conf files? -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atentamente, Jorge Gutiérrez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy on asterisk 1.6
I have read about that on asterisk 1.6, there will be a parameter o (Only
listen to audio coming from this channel), I have tried, but I still get
inbound and outbound audio from the spied channel.
Has anyone used this feature? Is it working? Is there any work-around?
I will like to only spy the outbound audio from a channel, I dont want to
hear the incomming audio of that channel.
I have used the following context:
[Conf]
exten = s,1,Answer
exten = s,2,Background(custom/menu_test)
exten = s,3,ChanSpy(,qoX)
exten = 1,1,Goto(Conf,s,2)
exten = 2,1,AGI(conf.php,${CALLERID(num)},${SPY_CHANNEL})
exten = 2,n,Goto(s,3)
exten = s,n,Goto(test2,s,1)
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Re: [asterisk-users] Fwd: Patton smartnode 463x (BRI) 25ms tail echo cancellation
Gaëtan, They are using as gateway to the pstn. In fact, they are remote gateways for a centralized callcenter: [pstn] «--» [BRI] «--» [internet] «--» [callcenter] Regards Jorge Mendoza Gaëtan Minet wrote: Thanks Are you using these to connect isdn phones to the voip or to as a gateway to the pstn for a voip system ? Kind regards Gaetan On 08/09/2009, at 19:40, Jorge Mendoza wrote: We have some installations with smartnode 4554, (same tail echo cancellation) without problems so far. Jorge Mendoza Gaëtan Minet wrote: Hi Is anybody using these ? Gaetan Begin forwarded message: *From: *Gaëtan Minet gminet...@mcit.be mailto:gminet...@mcit.be *Date: *Sat 22 Aug 2009 16:29:42 GMT+02:00 *To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject: **[asterisk-users] Patton smartnode 463x (BRI) 25ms tail echo cancellation* *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm however curious about their HW EC. I see in the datasheets that it only has 25ms tail per channel (pri are 128ms, but not BRI). Are some of you using these gateway and do your experience (many) echo problems on calls ? Our other alternative is to use sangoma cards that have 128ms HW EC and seem more stable overall, but it is yet a bit more expensive. Thanks for your feedback. Regards, Gaetan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Patton smartnode 463x (BRI) 25ms tail echo cancellation
Gaëtan, The echo arise at 4w/2w conversion point, normally at the far end. Try to call another phone number assigned to your BRI, i.e. call back to your Asterisk server. If you use a Polycom phone to initiate the call and another to receive the call, you have the perfect link: 4 wires with not 4w/2w conversion. Under this test, theoretically you must not have echo. If so, it is necessary to look elsewhere, at your Asterisk box maybe. Modifying the rx/tx gain is a good practice too. Best Regards Jorge Mendoza Gaëtan Minet wrote: Thanks ! We installed it in the interim and have a lot of calls with far-end echo :(. But it seems the solution could be to reduce the TX gain on our side (these are using polycom phones, and indeed I can see a big amplitude imbalance between tx/rx on a recording). It's under test, I hope it'll solve it. Kind regards, Gaetan On 09/09/2009, at 16:22, Jorge Mendoza wrote: Gaëtan, They are using as gateway to the pstn. In fact, they are remote gateways for a centralized callcenter: [pstn] «--» [BRI] «--» [internet] «--» [callcenter] Regards Jorge Mendoza Gaëtan Minet wrote: Thanks Are you using these to connect isdn phones to the voip or to as a gateway to the pstn for a voip system ? Kind regards Gaetan On 08/09/2009, at 19:40, Jorge Mendoza wrote: We have some installations with smartnode 4554, (same tail echo cancellation) without problems so far. Jorge Mendoza Gaëtan Minet wrote: Hi Is anybody using these ? Gaetan Begin forwarded message: *From: *Gaëtan Minet gminet...@mcit.be mailto:gminet- m...@mcit.be *Date: *Sat 22 Aug 2009 16:29:42 GMT+02:00 *To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject: **[asterisk-users] Patton smartnode 463x (BRI) 25ms tail echo cancellation* *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm however curious about their HW EC. I see in the datasheets that it only has 25ms tail per channel (pri are 128ms, but not BRI). Are some of you using these gateway and do your experience (many) echo problems on calls ? Our other alternative is to use sangoma cards that have 128ms HW EC and seem more stable overall, but it is yet a bit more expensive. Thanks for your feedback. Regards, Gaetan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Patton smartnode 463x (BRI) 25ms tail echo cancellation
We have some installations with smartnode 4554, (same tail echo cancellation) without problems so far. Jorge Mendoza Gaëtan Minet wrote: Hi Is anybody using these ? Gaetan Begin forwarded message: *From: *Gaëtan Minet gminet...@mcit.be mailto:gminet...@mcit.be *Date: *Sat 22 Aug 2009 16:29:42 GMT+02:00 *To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject: **[asterisk-users] Patton smartnode 463x (BRI) 25ms tail echo cancellation* *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm however curious about their HW EC. I see in the datasheets that it only has 25ms tail per channel (pri are 128ms, but not BRI). Are some of you using these gateway and do your experience (many) echo problems on calls ? Our other alternative is to use sangoma cards that have 128ms HW EC and seem more stable overall, but it is yet a bit more expensive. Thanks for your feedback. Regards, Gaetan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best ISDN BRI solutions?
We use Patton BRI gateways. No problems so far. If possible, we prefer to keep telephony interfaces out of Asterisk box. Regards Jorge Jaap Winius wrote: Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be? I might even be willing to try out a more expensive PRI card if I knew it also supported BRI: just as long as I would no longer have to worry about the software support for it -- for both Asterisk 1.4 and 1.6. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quenstion about asterisk
Hello fellows, I want to know if there´s a way to capture the numbers typed for a user; without waiting that the IVR finish or without predefine the numbers of digits. I´m going to explain you better, for example I want to know that a user typed 12345#, but I want that the user can type over IVR and don't predefine the numbers of digits X because the user should have the quantity the digits predefine. Thanks Elvis Jorge Cell: 809-706-8824 ETGTEL DOMINICANA La información contenida en este correo electrónico, así como los archivos anexos que pudiera incluir, es confidencial y únicamente para su destinatario. Si usted ha recibido este mensaje por ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
Could you give a example how I can do that?? Thanks - Original Message - From: Steve Howes st...@geekinter.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 17, 2009 10:34 AM Subject: Re: [asterisk-users] quenstion about asterisk On 17 Jul 2009, at 15:29, Elvis Jorge wrote: I want to know if there´s a way to capture the numbers typed for a user; without waiting that the IVR finish or without predefine the numbers of digits. I´m going to explain you better, for example I want to know that a user typed 12345#,but I want that the user can type over IVR and don't predefine the numbers of digits X because the user should have the quantity the digits predefine. Assuming you intend to use # as a terminator, just collect in a loop, 1 digit at a time until you get a hash.. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
The problem with read() is that I have to wait that a message that is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type
the quantity of digits predefine.
Could you give me other solution?
Thanks
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 17, 2009 11:06 AM
Subject: Re: [asterisk-users] quenstion about asterisk
On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
I want to know if there´s a way to capture the numbers typed for a
user; without waiting that the IVR finish or without predefine the
numbers of digits. I´m going to explain you better, for example I want
to know that a user typed 12345#,but I want that the user can type over
IVR and don't predefine the numbers of digits X because the user
should have the quantity the digits predefine.
On Fri, 17 Jul 2009, Steve Howes wrote:
Assuming you intend to use # as a terminator, just collect in a loop, 1
digit at a time until you get a hash..
Or, use read() or AGI's stream file.
For future reference, please take a look at:
http://www.catb.org/~esr/faqs/smart-questions.html#bespecific
There are many questions about Asterisk.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] is possible to sen sms with asterisk in Spain?
ESGLinux wrote: 2009/7/9 Anselm Martin Hoffmeister ans...@hoffmeister-online.de mailto:ans...@hoffmeister-online.de Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux: Hi all, I´m a beginner with asterisk and I want to know if with asterisk I can send sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with the operators...) Hi, the SMS code in Asterisk is - afaik - only for the landline type of SMS. It can behave as landline-SMS capable phone (like some of the Siemens Gigaset DECT devices, for example) and talk to a landline-SMS center that will for a certain charge forward short messages to mobile phones. It can also behave as landline-SMS center and talk to appropriate phones. As a background info, landline phones can recognize that a landline SMS center is calling them by caller ID (which must be programmed, many phones ship with the local companies' numbers preprogrammed) and will not ring the bell but silently answer the line. The message transfer works with 1200 baud modem-like analogue audio (even if the phone is an ISDN device) - you can watch the actual message bytes on the Asterisk CLI if you turn on debug, in some kind of simple protocol and some 8bit-to-7bit mapping. It cannot directly talk to mobile phones: short messages are transmitted out-of-band in the GSM networks, and the mobile operators will not allow you direct access there. After all, short messages make a hefty percentage of their income at a minimum percentage of infrastructure usage. The situation in Germany (and to my knowledge, in several other European states) is that you can connect to a premium-rate landline-SMS center and hand them a short message for relaying. As that is bound to cost hardly less than using a mobile phone directly, it is not at all interesting for me (ymmv). I prefer using one of those web-interface-to-sms providers (mine can be used with wget from scripts etc) and pay between 3 and 12 cents per message, depending on destination country and quality of service selection. They have been reliable for quite some time now, and I remember that landline-SMS was a little too fiddly for my taste. Regards Anselm ok thanks for your answer, I think your are right with the landline-SMS, Now my question changes to, how can I send a SMS to my cellular phone, what hardware, software, subcription to service or somthing else do I need? Thanks in advance ESG Take a look at: http://www.ozekisms.com/index.php?owpn=319 See Kannel as well: http://www.kannel.org/ Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and 4G
Hello AFAIK 4G will use IMS (IP Multimedia Subsystem) in the core network. Phones will use SIP as signaling. IMS is whay you are looking for. Greetings, -- Jorge Martínez López jorg...@gmail.com http://www.jorgeml.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IOS Interface
Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOS Interface
Brent Davidson wrote: Jorge Mendoza wrote: Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ You're comparing to apples to Orange. IOS is the Cisco operating system run by the Femtocells, not the protocol. I'm not familiar with Femtocells, but as far as I can tell (from reading wikipedia) they apparently can do SIP internally, but that is a more advance configuration and might require some additional software. Looks like they are more designed to what is called lub over IP which appears to be some sort of backhaul specification specific to Cellular / Wireless carrier technology. AFAIK, IOS interface for femtocell is a 3GPP2 specification. It is an application, no related to Cisco OS. Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] OpenBTS chat with David A. Burgess
Steve Totaro wrote: On Sat, Mar 21, 2009 at 4:17 PM, Steve Kennedy steve-aster...@gbnet.net wrote: On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. This looks like a great project, sorry I missed the call. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net Careful with this rabbit hole, it goes very deep and then logically branches in different directions then people become Of Interest, die In an Accident or Natural Causes, or disappear altogether. This, indeed, is a nice *technical* project. We follow up the project from some time. However from the practical point of view, has problems. The big one is regulation. In our country, a developing one, GSM bands are all licensed to the big operators, so you can not implemented a project like the OpenBTS, because you are no licensed. The big operators will never try to implement such kind of projects. And the regulators protects the big operators. Please, do not ask why. Regards Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is not designed for University large scale
IMHO when users scale up to such levels, Asterisk falls short, I made a c ouple large implementations and the best approach is using OpenSer as SIP engine (along with his own media proxy if required by your network schema) and use Asterisk as Vertical Services Provider, such as email, IVR, in general, expliding the benefits Asterisk overachieve, including TDM interconnection as well. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
See too: http://www.networkworld.com/news/2007/011907-mit-your-take.html?page=1 Jorge Mendoza Dean Collins wrote: Hi Visit, that's not correct - google Sam Houston University It's a pretty well known asterisk installation. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Li Sent: Tuesday, March 17, 2009 1:01 PM To: Yehavi Bourvine Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk is not designed for University with largeuser base? On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GEN-GEN and Manual Ring-Down (MRD)?
Raj Jain wrote: On Mon, Nov 10, 2008 at 6:56 AM, Tony Mountifield [EMAIL PROTECTED] wrote: Does anyone here know anything about GEN-GEN analogue circuits, also known as Manual Ring-Down (MRD)? Apparently they are widely used in Hoot'n'Holler systems for financial dealer-boards. I have been asked to try and interface to such circuits, and have been having great difficulty locating any specifications for the interface. Apparently, they are always-on 2-wire analogue circuits with no tip voltage or loop current, and on-demand superimposed ringing voltage in either direction for signalling (to do nothing more than get the remote end's attention). I was wondering whether it is possible to adapt an FXS or FXO port to operate in such a mode, but I'm not optimistic. Your understanding of MRD is correct (these are nailed-up connections with only ring-gen capability). I've personally not tried this w/ Asterisk FXS/FXO ports but If you can make it work that way pls. let us know. -- Raj Jain The MRD telephones uses local battery, that is the reason why they do not have loop current (central battery). Any adaptor to a FXS circuit is useless because there are not any signalling to indicate on/off hook. Just the initial manual ringing. Then working on FXS ports is almost impossible or very expensive. Another approach is to use E/M signalling. The audio channel could be open permanently and transmit the ringing over the E/M wires. You need a ringing detector and a ringing generator in both sides. Take care of isolate the audio channels from the ringing current. I do that many years ago on PCM muxes with E/M interfaces. The Multitech gateways have E/M interfaces, but never tested under this conditions. Obviously, the easy way is to use two standard sets working as hotline. Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agents log in afterhours
Hi all, I received a report of a client which stated that two of its agents are logging in to the queues when they actually arent there working. They appeared to be logged on all night. They thought they werent logging off correctly, but they checked one of them and he was following the procedure. Any ideas of what can be happening? Is there a way to prevent logins to queues afterhours? Thanks, Jorge Santiago Alanís Garza Innovación y Desarrollo mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue problem
Hi, I have 3 queues and they have the same weight. But one of the queues receives a lot of calls (much more than the other two) so people on that queue usually have to wait much more than the others. Is there a way to make asterisk determine the longest waiting call and give priority to that call, having the 3 queues (I know that if I had just one queue, this would be the natural behavior). Thanks, Jorge Santiago Alanís Garza Innovación y Desarrollo mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Tel: (81) .4044 Cel: (811) 243-6570 http://www.blocknetworks.com.mx/ www.blocknetworks.com.mx Av. Lázaro Cárdenas 4000, L-17 Col. Valle de las Brisas Monterrey, Nuevo León, CP 64790 Tel: +52 (81) 4044 Block Networks es una empresa certificada en ISO 9001:2000 Design, development, and sales of enterprise software and technology. La información contenida en este mensaje y sus anexos es de carácter privado y confidencial y para el uso exclusivo de la persona o institución a la cual ha sido enviado y para otros autorizados para recibirlo, por lo que no podrá distribuirse sin la autorización expresa del remitente. Si usted no es el destinatario a quien este mensaje fue dirigido o si no es un empleado responsable del envío de este mensaje al destinatario, se hace de su conocimiento que cualquier revisión, diseminación, distribución, copia u otro uso o acto realizado con base en o relacionado con el contenido de este mensaje y sus anexos está estrictamente prohibida y puede ser ilegal. Asimismo, el presente mensaje no representa la manifestación del consentimiento de ninguna de las partes, por lo que no genera derecho u obligación alguna para ambas sino hasta que sus representantes legales así lo manifiesten por escrito. Si usted ha recibido este comunicado y sus anexos por error, le solicitamos lo notifique inmediatamente al remitente respondiendo a este correo y borre el presente y sus anexos de su sistema sin conservar copia de los mismos. Gracias, Block Networks, S.A. de C.V. The information contained in this message and its attachments is private and confidential and is intended solely for the use of the individual or entity to whom it is addressed and others who are authorized to receive it; therefore, its distribution cannot be possible without authorization from the sender. If you are not the intended recipient or an employee responsible for delivering this message to the intended recipient, you are hereby notified that any revision, dissemination, distribution, copying or other use or action based upon or relative to the information contained in this message and its attachments is strictly prohibited and may be unlawful. You are also informed that the contents of this message shall not be considered as an agreement between the parties and shall not bind any of them until their attorneys decide to do so in writing. If you have received this message and its attachments by error, please immediately notify the sender by replying to this message and deleting it from your system without keeping a copy. Thank you. Block Networks, S.A. de C.V. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
Gordon Henderson wrote: On Sat, 11 Oct 2008, Jorge Mendoza wrote: I founded this behaviour in the past. When the CO provides reversal polarity and the FXO port is configured to ignore polarity events, then a reversal polarity could be detected as ringing if the hardware/software is not well designed or configured. So, if the CO provides polarity reversal, why not set answer and release supervision to yes? We need the flexability to answer either way... Here in the UK the (BT) exchange will do a polarity reversal to signal incoming CLI - it then send the CLI, *then* sends the ring signals, so answering on polarity reversal would be wrong. Answer supervision on reversal polarity applies only to outgoing calls, not incoming ones. They also do a random polarity reversal most nights too - some sort of automated line testing. Eg. from my home box: Oct 7 01:40:28 NOTICE[15581] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 9 03:53:06 NOTICE[19904] chan_zap.c: Got event 17 (Polarity Reversal)... Note the times... Are they just warning alarms or they starts phantom calls? Jorge Gordon Jorge Mendoza Jim Duda wrote: If by default Asterisk ignores all polarity events, then why does it cause the Dialplan to start? I did set answeronplarityswitch to no, however, I have had the problem occur once already, so, you suspicion might be correct. Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
I founded this behaviour in the past. When the CO provides reversal polarity and the FXO port is configured to ignore polarity events, then a reversal polarity could be detected as ringing if the hardware/software is not well designed or configured. So, if the CO provides polarity reversal, why not set answer and release supervision to yes? Jorge Mendoza Jim Duda wrote: If by default Asterisk ignores all polarity events, then why does it cause the Dialplan to start? I did set answeronplarityswitch to no, however, I have had the problem occur once already, so, you suspicion might be correct. Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI or PRI callerid
I managed to achieve that on a PRI line with the following: 1. On zapata.conf, for the PRI line channels, add facilityenable=yes usecallerid=yes usecallingpres=yes I do not known if these are all strictly required for anonymous calling, but it works for me. 2. On your extensions.conf, just prior do the Dial application invoke SetCallerPres(prohib_not_screened) 3. Your provider must also enable the apropriate functionalities on the PRI line. I believe they call IT CLIR (Calling Line Identification Restriction). Jorge Nunes Loic Didelot wrote: Hi, I try to get anonymous calling working on ZAP. But I am unsuccessful on PRI as well as on BRI. I tried all parameters from the application SetCallerPres(). Nothing worked. I even traced with my ISP and they told me that I am not sending any parameter to hide the callerid. I found on the internet articles and mailinglist posts dating from 2003 that did not really help me. Im on a recent asterisk 1.4 from SVN and using euroisdn. Can anyone help? Is there a way to sniff/trace zap channels in an asterisk independent way? Best regards, Loic Didelot. -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- PDMFC - Email: [EMAIL PROTECTED]; Web: http://www.pdmfc.com Phone: +351-213.572.029; Fax: +351-213.572.031 Address: Avenida Conde Valbom 30, 3 - 1050-068 Lisboa - Portugal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to force Asterisk 1.4 to use soxmix
You can use the variable MONITOR_EXEC in your extensions.conf to specify the shell command to be invoked by the Monitor application to mix the voice files. The shell command will be invoked with three command lines arguments appended: the file recording for the in leg (created by Asterisk); the file recording with the out leg (created by Asterisk); the path of the file where the mix result is supposed to be dumped to (i.e. the soxmix (or equivalent) outfile). Some examples in http://www.voip-info.org/wiki/view/Monitor+stereo-example Jorge Nunes Giorgio Incantalupo wrote: Hi Julien, the soxmix (or sox in Asterisk 1.4 as default choice) is used by Asterisk to record queues calls when you ask it to mix the in and out calls, so I do not have control on it. Asterisk 1.2.x uses soxmix while Asterisk 1.4.x uses sox instead but the command sox launched by Asterisk 1.4.x does not work on my Debian because the sox command version is 12.17.9, old but it is the most recent package available for Etch). The res_monitor.c code shows it is possible to specify which to use... soxmix or sox but I do not know how...I could use a #define HAVE_SOXMIX 1 before #ifdef HAVE_SOXMIX but I do not think is the right choice. Giorgio. Julien Claassen wrote: Hi! for which feature? I'm relatively new, but I guess, if it is dependent on some dialplan related stuff, you coudl always use a: System(soxmix Options) in the appropriate place. From what I've experienced upto now, you can setup a lot, which doesn't seem obvious. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- PDMFC - Email: [EMAIL PROTECTED]; Web: http://www.pdmfc.com Phone: +351-213.572.029; Fax: +351-213.572.031 Address: Avenida Conde Valbom 30, 3 - 1050-068 Lisboa - Portugal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Part of some calls does not get recorded
Hello, all. We have an Asterisk 1.4.17 installation and we have setup the dialplan such that all calls to/from a set of phones (SIP accounts) get recorded. We do this by ensuring the Monitor application gets invoked at the start of all calls. We also have canreinvite=no on the general section of our sip.conf. Things work as expected except in one case. In the following specific scenario part of a call does not get recorded. Assume three extensions (A, B and C) and the following sequence of events: 1. Extension A calls extension B and the call is answered on B. 2. Extension A puts the call to extension B on hold (B happily starts to hear the music-on-hold). 3. Extension A calls extension C and the call is answered on C. 4. Extension A transfers the call to extension C to extension B (an attended transfer). 5. Extension B now merrily talks with C and all is fine and dandy. Regarding the recording of the calls the results are these: 1. A recording file is created with the call from A to B. This recording ends at the moment when extension A transfered the call to extension C to extension B (yes, the last part of the recording is the music-on-hold part). 2. A recording file is created with the call from A to C. This recording also ends at the moment when extension A transfered the call to extension C to extension B. 3. The part of the call where B talks with C (that is, after extension A transfered the call to extension C to extension B) is not recorded anywhere. And this last item is the problem we have. We have the requirement that all calls to/from this set of phones must be recorded. But in this scenario part of the call does not get recorded. Could someone point me in a direction where I can start to solve this? Best regards and thanks in advance. Jorge Nunes -- PDMFC - Email: [EMAIL PROTECTED]; Web: http://www.pdmfc.com Phone: +351-213.572.029; Fax: +351-213.572.031 Address: Avenida Conde Valbom 30, 3 - 1050-068 Lisboa - Portugal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call-leg stays on MusicOnHold forever
Andreas, We can't help, but just to say that after 2 weeks of debugging, we have found yesterday that the one way audio experienced by the agents some times, is related to hold function. Jorge Mendoza Andreas Brodmann wrote: Hi I have a strange behaviour; perhaps someone who had a similar issue can help. I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco Call-Manager 6.1 cluster. Two phones/users from the Cisco environment call extensions on the Asterisk. Phone 1 / Call 1 is parked on the asterisk using: exten = xyz,1,Answer() exten = xyz,n,Set(PARKEXTENSION=555) exten = xyz,n,Park() Phone 2 / Call 2 is picking it up: exten = xyz,1,Answer() exten = xyz,n,ParkedCall(555) so far so good, they can talk to each other. Now if one of them presses Hold, Asterisk will: [Sep 5 14:16:05] VERBOSE[6351] logger.c: -- Started music on hold, class 'default', on SIP/10.16.17.162-081bb720 [Sep 5 14:16:05] VERBOSE[6351] logger.c: -- Stopped music on hold on SIP/10.16.17.162-081bb720 [Sep 5 14:16:05] VERBOSE[6351] logger.c: -- Started music on hold, class 'default', on SIP/10.16.17.162-081bb720 start - stop - start strange, but it works ... If the same user/phone now presses hold/resume so that they could talk to each other again Asterisk does: [Sep 5 14:16:07] VERBOSE[6351] logger.c: -- Stopped music on hold on SIP/10.16.17.162-081bb720 [Sep 5 14:16:07] VERBOSE[6351] logger.c: -- Started music on hold, class 'default', on SIP/10.16.17.162-081bb720 stops the music and starts it again ... now the guy who pressed hold at first can hear the other party, but the other party only hears music from Asterisk. Has anyone had a similar phenomenon? Regards, Andreas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri 1.4.5 priindication
Good morning, Into the libpri 1.4.5 announcement, it is stated that This version of libpri retains the ability to operate in this mode, but it is now a configurable option which defaults to being 'off'. The next releases of Asterisk will have configuration options to turn this behaviour on if the user desires Is this related to priindication? How I can to turn this option to on ? Which is the next release of Asterisk? Thanks Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancelation
Robor, The echo arise at the far end, where the 4W/2W conversion take place, not between the E1's. So, you should need an EC. Regards Jorge M. Robor Oghene wrote: Thanks Steve, Its an Ericsson and Siemens Switch within same room. On Thu, Jun 26, 2008 at 5:23 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Thu, Jun 26, 2008 at 12:17 PM, Robor Oghene [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, If am connecting a digium E1 card to a PSTN Switch in the same equipment room would I need an echo canceller? wouldnt the Switch handle echo cancellation for dial-in users? Responses would be appreciated. BR, Robor Switch is very generic and you give no real details. But, I would say you should be fine based on the tiny bit of info provided. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXW 4108 asterisk configuration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nelson Granados wrote: GXW 4108 asterisk configuration Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn’t work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yeah... it took me some digging but finally got it working by following the instructions provided by Grandstream. What is it exactly the problem? - -- Jorge Valdes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIWVGXvseWMACI1MYRAnZEAJ9saY6ogO8eEwmYqVCThwp0ODjGpACZAaoE ZZsys6XMvUGShDHmuESS4Mk= =en2Y -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with cisco 7970G [EMAIL PROTECTED]
Hi everyone This is the first time I post something here so I'm sorry about my English, I don't know how to write properly. Well, I've been working with Cisco 7960 telephones and my boss bought new ones , 7970G with SIP70.8-2-2SR3S firmware version, those work perfectly, but one of them has the SIP70.8.3.5S version, and this one doesn't connect to the server , I wanted to install the SIP70.8.2.2SR3S version, but I couldn't, is there anyone who knows how to do it? Many thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium PRI card hi-Z for sniffing?
Hi Tony, http://www.voicetronix.com.au/openpri.htm Never tested, though. We used the analogue boards for monitoring, so far. Jorge Tony Mountifield wrote: Does anyone know if the Digium PRI cards can be configured or modified to have a high-impedance input on the RX pair? I would be interested in this in order to build a bi-directional PRI audio sniffer using two E1/T1 ports per trunk to be monitored. Cheers Tony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about asterisk setup...
Hello guys.I'm new to asterisk, and I have a setup which * is running behind a firewall with 3 softphones installed on different computers, now the softphones can connect with no problem I can also make calls outside my network to ppl with ekiga and gizmo accounts, but my question is. can I receive calls with the setup I have from ppl with ekiga or gizmo accounts, without the need for a service provider?... if so could someone give me a hand getting this setup working? Thanks Jorge ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
I second Doug advice. Migrate to Asterisk asap. We have several Asterisk auto attendant integrated with Mitel, even billing using the Mitel's smdr. But voicemail is different. The COV card emulate a SS4 phone and receive information needed for a voice mail system. With FXO/FXS ports is not possible receive such information. Jorge Mendoza John covici wrote: Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines...
This is a well know issue in analogue trunks, called collisions or glare. As you say, more is the traffic more are probability of collisions. One trick to reduce this problem is to reverse the outgoing hunting group against the incoming hunt group. Jorge Mendoza Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answered Call marked as NO ANSWER
Raúl, From your conf file I guess the CO provide reversal polarity for answer supervision. Verify if for those numbers the CO revert the line polarity when callee answer. callprogress=no is a good test too. Jorge Raúl Gómez C. wrote: Hi list, I'm having problems transferring certain calls made by the attendant between the PSTN and to an internal extension. Although, transfers between the majority of the calls ends successfully. Debugin this, I've found that calls made to certain numbers (Telephony Providers), aren't detected as ANSWERED in the CDR, so they are not properly accounted (for billing), neither transferred to internals extensions. How can I solve this??? Is this a incompatibility issue between technologies??? Or just a config that I haven't made right??? Thanks in advance... My Setup: - Asterisk 1.4.17 - Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO) - Wanpipe 3.2.1 - Zaptel *MailScanner warning: numerical links are often malicious:* 1.4.7.1 http://1.4.7.1/ - Grandstream GXP-2000 Phones = *zaptel.conf* /# Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A400 [slot:4 bus:16 span:1] fxoks=1 fxoks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxsks=9 fxsks=10 fxsks=11 fxsks=12/ = *zapata.conf* /;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default ;usecallerid=yes ;hidecallerid=no callwaiting=yes usecallingpres=yes ;callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=0 callgroup=0 pickupgroup=1 callerid=Llamada Externa busydetect=yes busycount=4 callprogress=yes progzone=us hanguponpolarityswitch=yes immediate=no ;Sangoma A400 [slot:4 bus:16 span:1] context=watch group=1 signalling = fxo_ks channel = 1 context=fax group=1 signalling = fxo_ks channel = 2 context=from-zaptel group=0 signalling = fxs_ks channel = 3 context=from-zaptel group=0 signalling = fxs_ks channel = 4 context=from-zaptel group=0 signalling = fxs_ks channel = 5 context=from-zaptel group=0 signalling = fxs_ks channel = 6 context=from-zaptel group=2 signalling = fxs_ks channel = 7 context=from-zaptel group=2 signalling = fxs_ks channel = 8 context=from-zaptel group=3 signalling = fxs_ks channel = 9 context=from-zaptel group=4 signalling = fxs_ks channel = 10 context=from-zaptel group=5 signalling = fxs_ks channel = 11 context=from-zaptel group=6 signalling = fxs_ks channel = 12/ -- Nacho Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answered Call marked as NO ANSWER
Raúl, Callprogress is not reliable for call supervision. Sorry. For maximum reliability with callprogress, the tones and cadences send by the CO must match every well with the tones plan defined in your asterisk box. Probably the tones of the other telephone company, where the answer detection fail, are different or the cadences are different. Jorge Raúl Gómez C. wrote: Jorge, I think our telco doesn't provide disconnection supervision because I had to use callprogress, busydetect and busycount in order to properly disconnect a terminated call (and to avoid the infamous long message in the voicemail), so I think I can't disable the callprogress option. I will try to contact the telco provider of these numbers in order to ask them what kind of answer supervision they provide. Any other ideas??? Thanks again -- Raul Linux Counter #156439 On Fri, Feb 22, 2008 at 1:15 PM, Jorge Mendoza [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Raúl, From your conf file I guess the CO provide reversal polarity for answer supervision. Verify if for those numbers the CO revert the line polarity when callee answer. callprogress=no is a good test too. Jorge Raúl Gómez C. wrote: Hi list, I'm having problems transferring certain calls made by the attendant between the PSTN and to an internal extension. Although, transfers between the majority of the calls ends successfully. Debugging this, I've found that calls made to certain numbers (Telephony Providers), aren't detected as ANSWERED in the CDR, so they are not properly accounted (for billing), neither transferred to internals extensions. How can I solve this??? Is this a incompatibility issue between technologies??? Or just a config that I haven't made right??? Thanks in advance... My Setup: - Asterisk 1.4.17 - Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO) - Wanpipe 3.2.1 - Zaptel *MailScanner warning: numerical links are often malicious:* 1.4.7.1 http://1.4.7.1 - Grandstream GXP-2000 Phones ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
Yes, i have the same problem with att a few months ago, the problem is the acount (abonado) code, att need 2 and the code of unicall send 1, maybe the problem is the same for you, please post the debug unicall code. In this code, you can see the dial number, but if you see, the last digit is 1 On Feb 6, 2008 12:21 PM, Moises Silva [EMAIL PROTECTED] wrote: This is great news :) On Feb 6, 2008 10:56 AM, Carlos Chavez [EMAIL PROTECTED] wrote: On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote: Carlos, I have some spare time today in case you want me to check it. Is this your first time with Alestra? Thank you for the offer. Yes this is the first time I use Alestra for R2. I have another customer that uses them but with PRI and I do have some problems dialing certain numbers on that link. It turns out that there was a problem with their equipment but it took them almost 24 hours for them to admit it. It is now working properly. Calls now go in and out and for now I do not see any other problems. My list of tested providers for R2 in Mexico is now: Axtel, Alestra, Maxcom and Telmex. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI: calling an Unallocated Number
We have the following weird issue. When we call an unallocated number from asterisk through an E1/PRI euroisdn, the call disconnect with cause 31 (unspecified), This produce an Asterisk congestion message. If the same E1/PRI trunk is now connected to a Nortel BCM400, the call disconnect with cause 1 (unallocated number) which is correct, and the telco play the message the dialled number does not exist. All other calls work fine so far. Testing with priindication = outofband and priindication = inband give the same results. Any pointer please? Jorge Mendoza Our information: - OS Centos 5, 64 bits - Asterisk 1.4.13 - Zaptel 1.4.6 - libpri 1.4.1 - Wanpipe-3.2.1 (Sangoma A104DX) Attached are: - zapata.conf - zaptel.conf - debug_invalid_20071204 (a call to a unallocated number) T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (36) [ 00 01 01 49 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 036 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 45 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 034 P/F: 1 0 bytes of data -- ACKing all packets from 33 to (but not including) 34 -- 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 T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (36) [ 00 01 01 49 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 036 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 45 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 034 P/F: 1 0 bytes of data -- ACKing all packets from 33 to (but not including) 34 -- 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 T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (36) [ 00 01 01 49 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 036 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 45 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 034 P/F: 1 0 bytes of data -- ACKing all packets from 33 to (but not including) 34 -- 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 T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (36) [ 00 01 01 49 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 036 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 45 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 034 P/F: 1 0 bytes of data -- ACKing all packets from 33 to (but not including) 34 -- 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 -- Making new call for cr 32772 [ 00 01 44 48 08 02 00 04 05 04 03 80 90 a3 18 03 a9 83 81 6c 06 21 80 31 30 33 34 70 08 a1 34 32 38 37 30 35 36 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 034 0: 0 N(R): 036 P: 0 34 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a3] 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: A-Law (35) [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 06 21 80 31 30 33 34] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '1034' ] [70 08 a1 34 32 38 37 30 35 36] Called Number (len=10) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering
Re: [asterisk-users] Noise on FXS ports (Sangoma)
Stephen Bosch wrote: Hi, Jorge: Jorge Mendoza wrote: Never experienced with FXS modules on a PC with Asterisk. However we have experienced that kind of problems on legacy PBX without a good ground. If you replace the system with a analogue set and have not noise, then a ground current is generated in your system, probably originated at FXO side. Have you tested the PC isolated, with not lines and not switches? just the FXS calling the voicemail? No, I haven't gone that far yet, but it might be worth trying. One question I have: if this turned out to be the cause, what could I do to clean up the ground? There are so many elements -- the power supply ground, the telephone lines, the network cable ground, etc. -Stephen- Stephen, Good question. Finding ground problems is an art. First thing I should do: measuring your ground with respect of CO ground. With a voltmeter between the tip wire of your CO line (0 VDC ) and your local ground, voltage should be less than 5 VDC. Jorge ___ --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
Re: [asterisk-users] Noise on FXS ports (Sangoma)
Stephen Bosch wrote: Stephen Bosch wrote: Stephen Bosch wrote: Hi: I have a Sangoma A200 card installed in a server with two FXO modules and one FXS module. Analog sets connected to the FXS module have a squeaky static -- it's like static mixed with the sound of someone vigorously cleaning a window a few doors down. In other words, it's not a classic static noise, but it is noise, and it's distracting. Remote callers can hear this noise. I have, by turns: - Tested the line with the same analog sets plugged in at the demarcation point. No noise. (None of the SIP devices in this configuration have this noise for outgoing calls, so I'm sure it's got something to do with the FXS module). - Plugged the known good analog sets straight into the server - Moved the FXS module off the REMORA daughtercard to the main card - Replaced the FXS module with a new one - Run the server off of battery power, to see if the noise is garbage leaking in off the AC - Turned off the PoE midspan The next thing I'm going to try is turning off the switch it's plugged into. When that is done, I'll have done everything short of move the server to a different location. Maybe the power supply is generating this crap. Hmn. I'm going to test this hypothesis. Okay -- that didn't work. I swapped the power supply out with a better one, and even disconnected the extra ventilation fan. The noise is different but still there. To those with experience with FXS modules, I welcome your input. -Stephen- Never experienced with FXS modules on a PC with Asterisk. However we have experienced that kind of problems on legacy PBX without a good ground. If you replace the system with a analogue set and have not noise, then a ground current is generated in your system, probably originated at FXO side. Have you tested the PC isolated, with not lines and not switches? just the FXS calling the voicemail? Hope this helps Jorge ___ --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
Re: [asterisk-users] Alcatel - Asterisk setup
Following zapata.conf works for us, interconnecting Asterisk - BCM. Never tested with Alcatel though. Jorge Mendoza = Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=es context=from-zaptel signalling=pri_cpe switchtype=qsig rxwink=300 loadzone=pe defaultzone=pe channel = 1-15,17-31 ;for E1 callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=6 callprogress=yes faxdetect=incoming Vieri wrote: According to http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI the author had trouble with QSIG. It would be great if you could give me an extract of your zapata.conf in your successful QSIG setup. And any other tip for that matter. --- Jorge Mendoza [EMAIL PROTECTED] wrote: In my experience, many times Qsig is mandatory for interconnection between Asterisk and others PBX using PRI. Jorge Mendoza Vieri wrote: I'm having the same trouble when the Alcatel-Asterisk trunk has prefix meaning set to open routing number. I enabled overlapdial but still get the same behavior. When it's set to routing number Asterisk receives the full dialed number but it's limited to a maximum of 8 digits. Has anyone solved this open routing number issue that passes only the first digit and ignores the rest? --- Sahil Gupta [EMAIL PROTECTED] wrote: Hi, You need to enable overlapdial. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 On Tue, 29 May 2007, Carlos Hernandez wrote: Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into Alcatel) If / when we manage to get anything from Alcatel, we get just the first digit of the number the user is intending to call.. Asterisk expects the whole number at once, so it fails.. Most of the time we get nothing at all from Alcatel, we think something is missing, so Alcatel sees the link is down. Please let me know if you have done this type of work before. We are not wanting to involve the Alcatel people, unless really required. Is there any special way to set up zaptel/zapata so Alcatel detects the PRI to be operational? Is there any special way to receive the calls once the PRI is up? Right now asterisk is set with: pri_net Any information or hints will be greatly appreciated Thank you, Carlos NZ You snooze, you lose. Get messages ASAP with AutoCheck in the all-new Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_html.html ___ --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 ___ --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
Re: [asterisk-users] Alcatel - Asterisk setup
In my experience, many times Qsig is mandatory for interconnection between Asterisk and others PBX using PRI. Jorge Mendoza Vieri wrote: I'm having the same trouble when the Alcatel-Asterisk trunk has prefix meaning set to open routing number. I enabled overlapdial but still get the same behavior. When it's set to routing number Asterisk receives the full dialed number but it's limited to a maximum of 8 digits. Has anyone solved this open routing number issue that passes only the first digit and ignores the rest? --- Sahil Gupta [EMAIL PROTECTED] wrote: Hi, You need to enable overlapdial. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 On Tue, 29 May 2007, Carlos Hernandez wrote: Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into Alcatel) If / when we manage to get anything from Alcatel, we get just the first digit of the number the user is intending to call.. Asterisk expects the whole number at once, so it fails.. Most of the time we get nothing at all from Alcatel, we think something is missing, so Alcatel sees the link is down. Please let me know if you have done this type of work before. We are not wanting to involve the Alcatel people, unless really required. Is there any special way to set up zaptel/zapata so Alcatel detects the PRI to be operational? Is there any special way to receive the calls once the PRI is up? Right now asterisk is set with: pri_net Any information or hints will be greatly appreciated Thank you, Carlos NZ Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ --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 ___ --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
Re: [asterisk-users] SIP Echo
Try canreinvite=yes in order to confirm that CPU is not the problem. Jorge Mendoza Asterisk wrote: I tried with the ping ... all of the phones respond in ca. 0.3ms, so network seems to be OK. More than 90% of CPU on * box is idle even in peak times, so this shouldn't cause echoes either, right? Hmmm, so handset could be an issue, but did anyone ever experience any handset problems with Polycom IP SoundPoint 430 :-) ? I will check the headsets and any possibilities of acoustical echo. Besides that, if we rule out the network, and the CPU on the * box, is there anything else that could be causing echoes on internal SIP calls? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *David Gomillion *Sent:* Tuesday, May 22, 2007 3:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP Echo We experience echo too from time to time. It's usually headset-related, but not always. I ran a persistent ping on one of the phones, and we diagnosed a wiring problem with it. Other phones needed a new handset. The problem is that these problems need to be fixed on the phone NOT hearing echo. On 5/22/07, *Asterisk* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How could I check if bandwith or/and latency is causing it? If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way to gather more detailed info on SIP calls and latency? * box is connected to the 1Gb switch with 1Gb connection, and clients have 100Gb/s speed. CPU is 90% free in peak time, and there are 34 SIP hardphones connected to the * box. Thanks, Alex -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Alexandre VERNIOL Sent: Tuesday, May 22, 2007 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Echo Hi, Could be bandwith or/and latency ... Many causes... Alex Asterisk a écrit : Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --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
Re: [asterisk-users] asterisk and fax machine
Another solution: http://www.vikingtelecomsolutions.com/catalog/model_FAXJ-300.htm?sid=046EBF6027C7A0D38E77EAF75B184540pid=1209 Jorge aslay-pinwee wrote: Hi, Thank you very much. I will test your method ASLAY - Original Message - From: Thomas Artner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 21, 2007 6:49 PM Subject: Re: [asterisk-users] asterisk and fax machine Hi! Either the fax machine or the asterisk box has to pick up the call to know whether it is a fax or not. My solution is that I let asterisk pick up every call, and if it is a fax, then the call is forwarded to a fax-machine. If its a voice call, the call is forwarded to the phones. [incoming] exten = s,1,Answer() ;automatic answer for fax recognition exten = s,2,Wait(3);prevents ringing when it is a fax exten = s,3,Dial(Sip/21Sip/22Sip/25Sip/26,45,t) ;ring phones exten = s,4,Hangup ;hangup after 45 secondes ;is it a fax? then take it here! exten = fax,1,Dial(Zap/1) But this solution implies that asterisk picks up every call immediately. So the caller has to pay for the call before he can talk to you. tom aslay-pinwee wrote: Hi, I need to share my PSTN line with my Digium card together with my FAX machine. If fax coming in, will asterisk pick up the call or my fax machine pick up the call. How do I make asterisk not to answer the incoming fax and let my fax machine receive the fax. Similarly, how do I make my fax machine not to answer any voice call and let my asterisk answer.. Regards ASLAY ___ --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 ___ --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 ___ --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 ___ --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
Re: [asterisk-users] GSM Cards for Asterisk (UK)
We use instead sip gsm gateway, with 4 gsm modules from 2N. It should work in UK as well. Jorge Matt Brown wrote: Hi, I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. Do Digium make one ? as I am unable to find on their website or is it possible to compile the ztgsm parts into the current zaptel source ? *Junghanns if you are on list, please do not take the wrong way - the cards are fine, we use a QuadBri in our very own PBX - but it does mean we are having to run the experimental version from your website for asterisk 1.2, where as we would prefer to be using 1.4 :-) Regards Matt Brown ___ --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 ___ --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
Re: [asterisk-users] GXV-3000 IP Video Phone
There are a patch for Asterisk 1.2 allowing h.264. Please note as well that GXV-3000 last firmware works with H.263 too. Jorge Nitesh Divecha wrote: Thanks Dave, I did try with Asterisk 1.2 but it didn't work. The Video Phones came with H.264 Video Coder... Regards, Nitesh Andreas van dem Helge wrote: On 5/5/07, dave cantera [EMAIL PROTECTED] wrote: nitesh, you are correct. you need 1.4.x... daveC It is supposed to have H.263, which does work with 1.2.x: [general] ... videosupport=yes .. [video-enabled-sip-phone] ... canreinvite=no disallow=all allow=ulaw allow=h263 ... ___ --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 ___ --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 ___ --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
Re: [asterisk-users] OT: USB T1/E1 Interface?
http://www.gl.com/laptopt1.html Jorge Michael Collins wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Just for fun. I'm a telecom geek and having a USB T1 interface would be a fun toy to tinker with. Besides, it might lead to some useful products. -MC ___ --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 ___ --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
Re: [asterisk-users] echo cancellation and ztdummy
http://www.voip-info.org/wiki/view/Causes+of+Echo Rob Townley wrote: Please tell me what hybrid echo is? Where does it come from? Does it have something to do with analog vs T1 trunk lines? On 4/23/07, William Moore [EMAIL PROTECTED] wrote: On 4/23/07, Patrick Fortin [EMAIL PROTECTED] wrote: Are echo cancellation parameters useful when using the ztdummy driver and no physical card ? No. The echocan software and hardware only cancel hybrid echo. They do not cancel acoustic echo that would be generated by voip phones with bad speakerphones or bad headsets. ___ --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 ___ --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 ___ --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
Re: [asterisk-users] Asterisk PiX devices
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers Don E. Wisdom wrote: Hi All, Im just getting started in the asterisk world and im wondering if anyone can point me in the right direction towards getting asterisk working from my house to my asterisk server in my colocation facility. Thanks --Don ___ --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] tdm400 problem
Hi all I have a problem with an tdm400 with 2 modules 1 fxo 1 fxs it just doesnt load the fxs module i dunno why... zaptel.conf loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 fxsks=13 fxoks=14 zapata.conf [channels] language=es echocancel=yes context=from-pstn echocancel=yes echocancelwhenbridged=yes echotraining=500 rxgain=-4.0 txgain=-6.0 usecallerid=yes hidecallerid=no threewaycalling=yes ;; RDSI BRI switchtype = euroisdn signalling = bri_cpe group=0 channel = 1-2,4-5,7-8 #channel = 1-2,7-8 ;; FXO signalling=fxs_ks group=1 rxgain=1.0 txgain=-6.0 busydetect=yes channel = 13 ;; FXS signalling=fxo_ks group=2 context=from-internal channel = 14 -- dmesg Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Proslic Failed on Second Attempt to Calibrate Manually. (Try -DNO_CALIBRATION in Makefile) Module 1: FAILED FXS (FCC) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Registered tone zone 6 (Spain) -- [Jorge J. Boscán Etura] quando omni flunkus moritatus Linux 2.6.17 X86_64 running fc6, lu #137000 +34636029900 ___ --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
Re: [asterisk-users] Background / Invalid Extension through cell phone
Eric Eric ManxPower Wieling wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Doug Lytle wrote: John Congdon wrote: Hi, I have had an issue for a long time, and really just can't solve it. My boss and others seem to have a problem when they call into our asterisk phone system. It often takes 3-4 tries of entering an extension before the system gets it right. Below is my context that the call comes into, and some debugging from the asterisk console. You may want to add the following to the zapata.conf ; If you are having trouble with DTMF detection, you can relax the DTMF ; detection parameters. Relaxing them may make the DTMF detector more likely ; to have talkoff where DTMF is detected when it shouldn't be. ; ;relaxdtmf=yes I have found that relaxdtmf=yes has caused more problems than it fixes. In my experience problems with detecting DTMF on an FXO port can usually be fixed by playing with rxgain and txgain. What sort of problems have you seen it cause? I guess I could see hitting the wrong extension in rare cases. Anyways, relaxdtmf has worked wonders for me over T1s and analog lines (always seems to be cell phones that have issues, probably because of the GSM and radio distorts beyond the specs). It caused asterisk to see a single digit when two of the same digits were dialed in a row. So a user dialed 4415 and Asterisk saw 415. Remember that on all cell phones (except the analog ones) DTMF from the phone is sent out of band and so should not be distorted. Are you sure about that?. I think that the DTMF digits are send out of band before the answer supervision. After that, the DTMF digits are send in band if they are dialled from the keypad. When I call my IVR, the system answer and I dial other DTMF digits, only around 20% of calls succeed. However if I store the DTMF sequence in the cell phone (digits, pause, send, digits, etc.) 100% of calls succeed. Jorge Mendoza ___ --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
Re: [asterisk-users] mobility with asterisk
Take a look at Alberto Pastor's mail on this list at 2007/02/15: moving WiFi phone. +jm Alvaro Pacho wrote: Hello, I´m working testing every feature of asterisk in a lab. Now I am very interested in asterisk over network mobility environment. For example : when somebody is talking with his ip-phone ) and moving around a big enterprise, needing to change the ip-address (other AP) would it be possible in the minimum time to avoid loosing quality in the current call? I read this test http://lists.digium.com/pipermail/asterisk-dev/2006-December/025263.html but it´was written in December of 2006!! Were this ideas implemented? If you can help me with information about that please write me and I´ll test and give you my end result. Does anybody knows something about which is the best Cisco router to this mobility environment? Best regards, Pacho ___ --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 ___ --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] Siemens HiPATH 3700 with Asterisk
Hi, I will like to know if anyone would guide me about how I can to interconnect one SIEMENS HiPATH 3700 with Asterisk. HiPATH have VoIP card and my idea is to do one un IP trunk between them so we would to transfer calls and services (voicemail, IVR,..) between both. We havent PRI ports unused in HiPATH so cheapest method of interconnection is one IP trunk. Any help or comment about will be interesting. Thnks in advance ___ --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
Re: [asterisk-users] upgrading from A101 to....A102
Hi Jeremy, We had D channels problems with A102De (A102 with HWEC and PCI-Express version), and it was solved from Sangoma changing one parameter in wanpipe.conf. We have HP server too in this installation. Our problem with D-channel was when wanted use only half-E1 channels (really we continue having 15 channels up from telco), and we wanted limit them in wanpipe config. Here show you our wanpipe.conf: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 14 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES Our change was set ACTIVE_CH = ALL and every sync problems with telco about D-channels was solved. Hope this helps you. Regards On 23/2/07 17:16, Porier, Jeremy M. [EMAIL PROTECTED] wrote: We're having a lot of D channel problems with the pci-e on HP servers. Going to PCI fixed the problem. Sangoma is aware of the problem and is using one of our servers to work toward a solution. -Jeremy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, February 22, 2007 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] upgrading from A101 toA102 Any benefit on getting the PCI Express version? Bill ___ --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 ___ --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
Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
Daniel Kocher wrote: Daniel Kocher wrote: I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher It kind of depends on what you're trying to accomplish. What do you want to be able to do with this connection? -Dave I would like to use the * as VoIP Gateway. Something like that: A user takes off a phone on a Avaya extension and dials for example 8 to reach the CO Port. Then Asterisk answers and sends a dial tone. The user dails a numer and Asterisk is doing the rest! (Sending the call to an SIP or IAX Provider) Yes, it works fine like that. We have several systems using * as a gateway. Jorge Mendoza ___ --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
Re: [asterisk-users] Billing pulses
Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Jorge Mendoza Stefano Corsi wrote: Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: frequency 12 kHz ± 1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ± 25 ms pause duration 180 ms period ... 300 ms Does someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming? Thanks Stefano ___ --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 ___ --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
Re: [asterisk-users] Billing pulses
All digital lines (BRI or PRI) provides answer and release supervision. The drivers will send to * this information, and this information will be registered into the CDR automatically. You only need setup your billing system. As said before you do not need to intercept the billing pulse. Jorge Mendoza Stefano Corsi wrote: At 16.22 07/02/2007, you wrote: Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Yes, it's strange. But I find no mention on answer supervision in the NT1Plus manual (NT1Plus is the hardware device the Telco installs when you ask for an ISDN line). Where should I ask for answer supervision? The Telco? That sounds very difficult in Italy... they have no technical call centers. Almost only sales. But if the line should provide those analog billing pulses... do you think could be possible to intercept them? Rgds Stefano ___ --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
Re: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
Noah is correct. We will install a trial system with 11 AP. The WiFi terminal will hold a conversation when moving between APs. Initial tests with Hitachi IP5000 were ok. We need to test as well PDA and cell/WiFi phones. Jorge Mendoza Noah Miller wrote: Roaming is irrelevant in VOIP. You just need a fairly good wifi connection. I don't think they mean roaming in traditional cell-phone terms. I think they mean moving between different Access Points on a single WiFi network. Judging by the reports in this thread, some Wifi phones do this better than others. ___ --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 ___ --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
Re: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
Have you tested the Ipaq or Asus with softphones in a roaming environment? Jorge Mendoza Vernier Umali wrote: The best experience I had in using a wifi handset to connect to asterisk is a windows mobile based PDA. I had the priviledge of testing a few phones in our company to connect via VOIP. I do not have any luck using nokia E61 (doesn't register and keeps on hanging). I would think it's the same with all wifi enabled nokias. I used an Ipaq 6900 series and Asus P55 and both worked well with SIP (SJphone) and IAX (PPCIAX). For me, this would be better since I will not be carrying a phone, a PDA and a VOIP phone. It's all in one device. ___ --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 ___ --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
Re: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other
Singer,
Assuming that you have no issues with firewalls in the path regarding
the rtp ports, or hardware/firmware problems, take a look at this patch:
http://www.sineapps.com/news.php?rssid=1019
Please take note if * does not receive rtp packets for any reason, it
does not send either.
Jorge
Singer Wang wrote:
I have bindaddr=0.0.0.0 in my sip.conf; what my major problem is that it
only happens 5-8% of the time..
On Wed, 2006-12-06 at 09:56 -0500, Ed Nuñez wrote:
If you use both the public and private interfaces for VoIP in the Asterisk
Server, make sure you don't specify one of them for the binding in sip.conf
Example
bindaddr=0.0.0.0
will allow SIP traffic on any of your interfaces.
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Singer Wang
Sent: Tuesday, December 05, 2006 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] problem with asterisk - calls where both sidescannot
hear each other
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
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Re: [asterisk-users] snom subscriptions issue on WRT
We had the same problem with WRT54G with no Linksys Linux firmware. At that time the problem was WRT54G modified the devices IP address, i.e. Asterisk received the WRT54G IP address instead of device address. Solution was selecting NAT=yes. Hope this help Jorge tommaso.carrara wrote: Hi, I've just installed asterisk 1.2.1 on my openwrt distro ( I own a WRT54GL by Linksys ) . No problem by now, but I can see that my 3 snom 320, once they started they send subscriptions to asterisk, and I can see that running: sip show subscriptions But, after one hour about, OR when I do asterisk reload , asterisk losts all th snom subscriptions. Someone can help me please? Thanks ___ --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 ___ --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
Re: [asterisk-users] asterisk and norstar
Gustavo,
Glad to help.
Gustavo, Linux and Asterisk are tools for implementing a telephony
system, so you must to know telephony basics first. Fortunately Asterisk
will force you to learn telephony!!.
Regarding transfers, see the following scenario. A calls B and B
transfer the call to C.
Blind transfer.- B dial C number and hangup immediately. Doesn't matter
if C answer, if C is busy or if C doesn't answer.
Supervised transfer.- B dial C number and stay in the middle to see if C
answer, is busy or doesn't answer. Then B will take actions depending on
C state.
Regarding topology, yes we have installed auto attendant in many
customers and voicemail, only tested on our old Mitel SX-100 at lab. At
our office we use Asterisk from his early ours!.
Hope this help.
Jorge Mendoza
Gustavo Berman wrote:
Hello Jorge, and thanks for the answers, but:
I don't understand what is a blind transfer and a supervised transfer.
I mean, in the topology:
- pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk
An incoming call from the pstn line is forwarded by the norstar to
extension 123 were asterisk is.
So asterisk answers the call and play a background message for the
caller.
But when the user enter the extension number what do we have to do?
I tried with:
Hook flash version:
exten = _XXX,1,Flash() ;do a hook flash (like pressing FUNCTION in
meridian phone)
exten = _XXX,2,SendDTMF(*70w${EXTEN},250) ;sends the code for
transfer plus the extension
exten = _XXX,3,Hangup()
In this version I can transfer the call using the same channel (zap/1)
but didn't find a way for voicemail if the call is unanswered or is busy.
Also if its unanswered the call is returned to the extension were
asterisk is.
Dial version:
exten = _XXX,1,Dial(ZAP/1/${EXTEN})
It says the channel is busy.
I think that with this version I can have a dialstatus for sending to
voicemail
So, a couple of questions:
What is a blind and a supervised transfer? (cannot find it in the
norstar manual)
Do you have and use this topology? if so, how do you do it?
Thanks for the help!!
(I'm a linux sysadmin and never before worked with telephones system)
--
Gustavo Berman
Sysadmin
Depto. Informatica
Universidad Nacional del Comahue
Centro Regional Universitario Bariloche
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Re: [asterisk-users] WIFI phones on asterisk
Andrew, Could you explain what problems do you have with Hitachi 5000?. We have carried out extensive tests with Hitachi 5000 at customer location who is planning to install more than 120 wifi phones. It is a mining company at 4200 mts altitude, covering the mining camp, an small village and the road within. Test were coverage, roaming, battery life, easy to use and Nortel-Asterisk integration. We had good success. Radios were Proxim 4000 and 700. For us is very important if you point out a problem. Are we missing something?. Jorge Mendoza Andrew Joakimsen wrote: I am surprised that you have had good success perhaps you haven't done proper testing? On 11/10/06, *Jerry Geis* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have used WIP300, hitachi 5000 wireless phones on asterisk and have had good success. However, I am looking for a WIFI phone with integrated belt clip. Has anyone found any? I have tried after market clips and holders and those just don't work. THanks for sharing if someone has found something that works with asterisk. Jerry ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --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
Re: [asterisk-users] asterisk and norstar
Hi Gustavo, Auto attendant is easy, voicemail I don't think so (there are not extension information when call is back to Asterisk). We use the following topology: - pstn line - norstar (ext 123) - (fxo) asterisk Jorge Mendoza Gustavo Berman wrote: Hi there! We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension. We are in Argentina, so buying a star talk is out of the question, there is no selling of that in here. So, we want to use * as an auto attendant and voicemail for our 50 extensions. Is there anybody who has done that? What topology do we have to use? : 1) pstn line - (fxo) asterisk (fxs) - norstar or 2) pstn line - (fxo/1) asterisk (fxo/2) - ata - (ext. 123) norstar or 3) pstn line - norstar (ext. 123) - ata/1 - (fxo/1) asterisk (fxo/2) - ata/2 - ( ext.321) norstar or 4) pstnl line - norstar (ext. 123) - ata - (fxo) asterisk Any help please? I'm not a telephone systems specialist! Thanks! -- Gustavo Berman Sysadmin Depto. Informatica Universidad Nacional del Comahue Centro Regional Universitario Bariloche ___ --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 ___ --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
Re: [asterisk-users] asterisk and norstar
Hi Gustavo, I correct myself. Voicemail is possible if you make a supervised transfer (I was talking about blind transfer). Sorry for my too fast response. Jorge Mendoza === Hi Gustavo, Auto attendant is easy, voicemail I don't think so (there are not extension information when call is back to Asterisk). We use the following topology: - pstn line - norstar (ext 123) - (fxo) asterisk Jorge Mendoza Gustavo Berman wrote: Hi there! We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension. We are in Argentina, so buying a star talk is out of the question, there is no selling of that in here. So, we want to use * as an auto attendant and voicemail for our 50 extensions. Is there anybody who has done that? What topology do we have to use? : 1) pstn line - (fxo) asterisk (fxs) - norstar or 2) pstn line - (fxo/1) asterisk (fxo/2) - ata - (ext. 123) norstar or 3) pstn line - norstar (ext. 123) - ata/1 - (fxo/1) asterisk (fxo/2) - ata/2 - ( ext.321) norstar or 4) pstnl line - norstar (ext. 123) - ata - (fxo) asterisk Any help please? I'm not a telephone systems specialist! Thanks! -- Gustavo Berman Sysadmin Depto. Informatica Universidad Nacional del Comahue Centro Regional Universitario Bariloche ___ --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 ___ --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 1.2.x and video
Hi, I would like to know which is the lasted Asterisk 1.2.x version (branch or trunk) for video support with h264 codec, and where I can downloaded it. Thank You Jorge Mendoza ___ --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 Solaris
Has anybody tried running Asterisk on Solaris on a SUN SparcStation ? Or maybe the alternative of running Asterisk on a Linux Distro on a SUN SparcStation? I am asked to do this but I think it's almost impossible work to make it happen. Regards, Jorge A. ___ --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
Re: [asterisk-users] VOIP Bandwidth questions
Capacity is planned using Erlang Formulae which is a medium complexity statistical model mainly used for voice communications trunk occupation and switching capacity. Some idea of bandwith usage might be obtained using the simple calculators at www.voipcalculator.com Regards, Jorge A. Erick Perez wrote: This one will surely heat up. Usually the telcos have to calculate the subscribers vs telco capacity. I use simple figures, so extrapolate this to millions of customers, millions of lines, peak amount of calls at any given time of the day and of course houndreds,thousands of millions of dollars in equipment. For example: Telco A has 100 subscribers to his phone service in a city (home and business), so he needs to ask himself a- Will the telco buy a switch that can handle 100 calls simultaneously? So he can provide service to his subscribers 100% of the time at any time of the day even during riots,panic,flood,etc? b- Or will the telco go for a balance and guess that at the peak time of the day he will have 75 simultaneous call, so he goes out and buy a switch that handles 75-80 calls at the same time? c- how many trunks will the Telco have to talk to other telcos? So telco in City A can communicate with Telco in city B (or even in the same city)? International voice providers suffer from this kind of problem. Some sell plastic cards with a local phone number and a pin so you call them to call to other cities/countries but that cheap voice provider has, let's say, ten thousand long distance lines and ten thousand local phone numbers, but they sell 100k plastic cards a month with a peak usage 3 times every ten days of 12thousand lines? obviously 2 thousand callers wont get connected (only 3 times every ten days in a specific time range) but the other 7 days the peak usage is 10thousand calls? Every ten days the provider try to connect 106k calls but fail to connect 6k calls, that's 6% failure rate every ten days (100% in a 7 days period and 98% in those 3 days). Can you live with that failure ratio? that's up to you. I don't work for a Telco, but a Telco may apply the dialup-internet rule (and they live happy with it) for 30subscribers-to-1line home users and 10(or 5)subscribers-to-1line for business. (correct me if I'm wrong please it will be nice to know real figures). So apply the same rule to you VoIP hosting. -What codec will you use? let say g711 and let's say it uses 100kilobits per leg. -How many subscribers will you have in a 6 month period? 500 -So to provide all of them with service you will need 48Megabits of bandwith all the time just to connect to your Telco equipments. - But you decide that you analyzed the usage patterns of your service and you will have only 125 subscribers calling other 125 subscribers (this is called On-Net) at peak time every day at 6pm (rush hour). So, go out and buy 24mbits of bandwidth only. - But you suddenly have the option to hire burst IP service where your IP carrier can provide you with more bandwidth if your usage starts to rise in any given time of the day. So you calculate again that your minimum constant usage at any time of the day is 40 users On-Net, so go out and buy 5mbits (for a total of 50 calls) of bandwidth with burst IP enabled from 6pm to 8pm of 48mbits (or 24mbits). This scenario is only subscriberyour_companysubscriber. you also need to calculate subscriber--your_companyother_telcos And the last but most important question is: how much money do you have to burn on this? 100% Uptime full-service, Top Carrier Class performance (and even they get busy sometimes)? or almost perfect service with the once-in-awhile glitch of we're sorry all circuits are busy, please try again. Hope this helps, How many times (at least in my country) haven't you suffered from Im sorry all circuits are busy, please try again during christmas midnight, new years eve, election days or similar behaviors that cause massive amounts of calls being initiated and received? So the answer to your question On 11/2/06, mail-lists [EMAIL PROTECTED] wrote: Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way
[asterisk-users] Asterisk 1.4beta3 and Asterisk Manager API Action: ExtensionState
Hi, We are testing Asterisk 1.4beta3 (same problem with beta2) and have the following problem. After some time (depending on traffic) the extensions are detected as busy. We use FreePBX, and the sequence to detect extension available is: dialparty.agi - is_ext_avail - ExtensionState - StatusCode When Asterisk is in good state the StatusCode is 0. In wrong state the Status is 16! The Status Codes list is: Status Codes -1 = Extension not found 0 = Idle 1 = In Use 4 = Unavailable 8 = Ringing A reboot of Asterisk, clear the wrong condition. If we go back to Asterisk 1.2.7 (for instance)all works fine. What we are doing wrong?. Thank You for your time. Jorge Mendoza ___ --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
Re: [asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???
See: http://www.voip-info.org/wiki/view/Asterisk+SS7 Jorge Mendoza Jay R. Ashworth wrote: On Thu, Sep 21, 2006 at 05:15:36PM -0600, MF wrote: Hi I need to connect at least 2 (and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done with Digium cards? if not, which box could I use to convert from SS7 to isdn, (could anyone please recomend one of these boxes, and let me know a ballpark price figure) If you're connecting to a carrier's SS7 network, I'm pretty sure you need to be using carrier-lab-approved hardware -- and very probably software -- to do it. Things may have changed since, oh, 5 or 6 years ago when I last paid any close attention to SS7, but last I heard, the ingress ports to that network are not filtered enough for them to let just anyone hook up to them. That said, those links *used* to be V.35 off the terminal equipment; I don't know whether they're using T-spans for them now, but even if they are, I suspect you might need custom *drivers*, not just custom app-level software. But IANASS7E. Cheers, -- jra ___ --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
Re: [asterisk-users] Hangup on Panasonic KX-TEM824
No way if you are using fxs on panasonic and fxo on *. jorge [EMAIL PROTECTED] wrote: I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect HANGUP from this. Can anyone help me to get it work. Thanks! ___ --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 ___ --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] Please help with a telular mod. SX5e
Hi I have 5 telular mod SX5e https://www.telular.com/v2/html/products/product_display.asp?productID=94 is a great stuff, but i have a extrange problem with asterisk. Sometimes the sound is ugly or choppy. The telular alone work fine all the time For example if i made 5 calls from asterisk to gsm network, but 2 or 3 calls the sound is really bad but only in the side of asterisk. I try all the echo cancellers but nothing work. I have 2 setups sipphone == asterisk with channel bank == telular SX5e == GSM network sipphone == asterisk with wctdm 4FXO == telular SX5e == GSM networkIn both case is the same. Any idea? do you have a similiar problem?Thanks ___ --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] Articulation Palm client and Asterisk
Hello, Has anybody configured Asterisk and the Articulation palm client to work ? I can make calls but I cannot make it register to receive calls. It does not register to the box. There are so few parameters that I think Asterisk sip.conf must be changed somewhat. I do not pass any parameters here because my box works perfectly with polycom, grandstream and linksys/sipura, and I know what to touch. The articulation software has only SERVER,DOMAIN, DISPLAY NAME, USER, PASSWORD, codecs are configured correctly (it only supports G711u and GSM), and I configured SERVER=DOMAIN (ip address) since it does not try to register until DOMAIN has something in i, Regards, Jorge Alayon ___ --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
Re: [asterisk-users] PRI and Asterisk
Never tested Redfone box. Digium and Sangoma cards works fine for me. Jorge Mendoza Julian Varanini wrote: Hi Everyone Any opinions on this? Thanks Julian From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Tue, 18 Jul 2006 01:29:57 + Subject: [asterisk-users] PRI and Asterisk Hi All, I am planning to order a PRI and would like to know your opinions on a devices like the Redfone redbridge. Basically any PRI to Asterisk interface that has worked well for you. Thanks, Julian ___ --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 ___ --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
Re: [asterisk-users] E1 for Voice and Data with MFC/R2
Hi a few months ago i use a digium card in a E1 with 10 voice channels an 10 in data, i use a gentoo distro with kernel 2.6, if you can tellme more info for example the channels, the type of card, kernel, etc maybe i can help you On 8/7/06, Moises Silva [EMAIL PROTECTED] wrote: I just found this spec page where it describes very well what can bedone with this sangoma cards. But for the few lines ive read, may bewith sangoma you cannot divide a single E1 link using some channelswith data and others with voice, in theory, again, that should be possible with Digiums cards. You can confirm that directly with Digiumsupport, if yo do that, please post here the response :)The sangoma specs page of the card I have used is: http://www.sangoma.com/datasheets/p_aft-et1-specsI used the 2 ports card, A102.RegardsOn 8/7/06, Carlos Chavez [EMAIL PROTECTED] wrote: On Mon, 2006-08-07 at 14:18 -0500, Moises Silva wrote: I have done something similar with Avantel, but not sharing channels in the same link. I received 1 E1 line for voice, and other E1 line for Internet, but in theory sharing channelsshould not be a problem. I could not make HDLC work with the kernel HDLC generic software driver and Digiums cards, so I used Sangoma's instead, and worked perfectly. Sangoma have their own kernel drivers for as far as i remember. If you want a quick installation, i would suggest you to use Sangoma cards. In fact Digium provided us with very little support for HDLC. A plus for Sangoma is that theyhave a simple console graphical user interface for configuration. Any specific card model for sangoma?Someone was telling me that they have to be the models that end in c. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQBE16IqVhw7eWImqUMRAigjAKCXUOjDzSnD/j3bbrs1afl7YDntjwCeKA3m TxEDUR7dIPsemkYY1qIRzZ4= =W85d -END PGP SIGNATURE- ___ --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 --Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
Even if he has r in the dial plan? Jorge C F wrote: Then you have something wrong some other place, if you are using an FXO card then asterisk is not even giving you the ring, the panasonic is. On 8/2/06, Pablo Mora [EMAIL PROTECTED] wrote: I think still didn't explain me clearly… The problem is when I dial 0, in this case the asterisk take Zap (connected directly to ext 200 from Panasonic), Panasonic gives tone, dial another extension (ie 100), the extension rings but when answer the phone asterisk keeps ringing… it doesn't detect when you pick up the phone. ___ --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 ___ --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 ___ --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
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo, according to description I assume that you have an FXO at * connected to an FXS port at Panasonic. If this is correct, could you replace Asterisk by a telephone and see if it is possible to make call to Ext1? Jorge Pablo Mora wrote: /Ok Ok, the figure doesn’t help./ / / /Here we go again…/ / / / / / - -- --- --/ /| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN |/ / - -- --- --/ / | |/ /Ext1 Ext2/ / / / / /Here is my dialplan/ / / /[incoming]/ /exten = s,1,Answer/ /exten = s,2,Background(prueba-pbx)/ /exten = s,3,Set(TIMEOUT(response)=5)/ /exten = 1001,1,Dial,SIP/1001|20/ /exten = 1001,2,Hangup/ /exten = 1001,102,Congestion,3/ /exten = 1002,1,Dial,SIP/1002|20/ /exten = 1002,2,Hangup/ /exten = 1002,102,Congestion,3/ / / /[sip]/ /include = outgoing/ /exten = 1001,1,Dial(SIP/1001,20)/ /exten = 1001,2,Hangup/ /exten = 1001,102,Congestion,3/ /exten = 1002,1,Dial(SIP/1002,20)/ /exten = 1002,2,Hangup/ /exten = 1002,102,Congestion,3/ / / /[outgoing]/ /exten = 0,1,Dial,Zap/g1/ /exten = 0,2,Congestion/ /exten = 0,102,Congestion/ / / /exten = 9,1,Dial,Zap/g1/9/ /exten = 9,2,Congestion/ /exten = 9,102,Congestion/ / / /When I make a call from PSTN to SIP, first Answer the Panasonic, after this I digit an Extension and the call goes to asterisk, then I dial to sip and the call goes on successfully. / /When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the call goes to asterisk, then I dial to sip and the call goes on./ /When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap sending 9 to get PSTN line, the dial the PSTN number and the call goes on./ /When I make a call from SIP to Ext1 (Ext2… ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything./ / / /Your help will be appreciated./ / / / / / / ___ --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 ___ --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
Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk
I want the real caller ID to be sent to Asterisk, which means I don't want the ATA to register. The badly written Sipura docs aren't clear about how to do this. Anyone set this up? I am having the same problem... Cheers, Jorge Mauricio -- blog http://djmaucom.blogspot.com http://jmauricio.blogspot.com /blog ___ --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
