[Asterisk-Users] Re: Dell PowerEdge SC420 interrupt issue
We're using an SC420 and using BRI with a quadbri Junganns card, with IAX softphones and one hardphone. Working well except that we sometimes get dropped connections between IAX and the server with a max retries exceed message, which comes from the chan_iax driver code. The BRI side of things looks like it is fine. I had been thinking it might be a network issue but now wonder if it is an interrupt or other background process issue causing a timeout on the Dell - hence my post as it might be the same cause as yours. We're about to concentrate on this hypothesis. If it is then it could perhaps be due to: 1. Linux - we're running Debian 2.6.8 2. Something in the firmware - we have twin SATA drives, though not mirrored as we had orginally expected. 3. E-mail background process. Doubt it as it is only used for voicemail messages. 4. Windows networking/SAMBA share. We only use this for configuring the conf files from windows and backing up configuration etc. 5. Other background process. Perhaps moh? We're using madplay though I've just checked and noticed a few perhaps rogue mpg123 processes. 6. Overloading? We're only a 10 person office so figure the SC420 with 2.6 G Celeron should be enough. So no solutions here for you but using same platform with what looks like a timeout/background process type issue. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Dell PowerEdge SC420 interrupt issue
On Fri, 17 Jun 2005, Paul Redstone wrote: We're using an SC420 and using BRI with a quadbri Junganns card, with IAX softphones and one hardphone. Working well except that we sometimes get dropped connections between IAX and the server with a max retries exceed message, which comes from the chan_iax driver code. The BRI side of things looks like it is fine. I had been thinking it might be a network issue but now wonder if it is an interrupt or other background process issue causing a timeout on the Dell - hence my post as it might be the same cause as yours. We're about to concentrate on this hypothesis. If it is then it could perhaps be due to: You could try running Asterisk with realtime privileges and see if it makes any difference. This will make the userland code preempt any other userland code. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inbound agent recording filename
We upgraded to cvs-head a couple of days ago, but haven't changed the config files at all. Prior to the upgrade, all inbound calls to our call queues were recorded, and the filename was like agent-6043-1109793719-24472.gsm. After the upgrade, the filename is now 1109793719-24472.gsm What do I need to change, (or can I) to get back to the old naming convention - it was really useful to see the agent number as part of the filename. Help :) Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID
Hi, My telco provider is SBC. I think that they use FSK to transmit caller ID. How can I set-up Asterisk so that I can see caller ID on incoming calls. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Calin Serbanescu Sent: Wednesday, June 15, 2005 19:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Caller ID What standard does your telco send the caller-id in ? ETSI FSK, bellcore... ? On Wed, 2005-06-15 at 16:26 +0200, Stojan Sljivic - GDS wrote: Hi Juan, I have Caller Id service enabled. When I connect the line to the phone I see the caller Id on the phone's display. I have callerid=asreceived. I have also played with various combinations of cidsignalling and cidstart, but with no success. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Manuel Coronado Z. Sent: Wednesday, June 15, 2005 15:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Caller ID Hi, First, you must ensure that the your Telco is sending you the caller ID with incoming calls. In some countries this is an aditional service you have to pay for, upon request. If you already have the service from the Telco, check in your zapata.conf that you have callerid=asreceived on your channels and group definitions. Hope this will help. Regards, Juan Manuel Coronado Z. On mar, 2005-06-14 at 14:50 +0200, Stojan Sljivic - GDS wrote: Hi, I'm using TDM04B and Asterisk 1.0.5. How can I setup the Asterisk so that I get caller ID? I do not get caller ID currently. Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip INFO DTMF over satellite
Hi all Am experiencing a very weird behaviour with sip info DTMF . I have an * box which is a satellite hop away. When I make a call from a Grandstream call set to send DTMF thru sip info, am able to navigate through the * menus very well. meaning that the DTMF is being recieved very well. The problem is when I connect to an external sip server that has an IVR. The external sip server does not seem to recieve the DTMF from the GS phone. From an * server on the LAN connecting to the external sip server, the DTMF are recieved very well on the external server. I read something about the 250ms of the Sip INfo Mode of DTMF relay but its very shallow and have no idea on how to change the settings on Asterisk. http://www.voip-info.org/tiki-index.php?page=SIP%20Info%20DTMF Note: dtmf-relay or dtmf are not yet IANA registered application mime types Cisco uses SIP INFO for DTMF relay: See http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftinfo.htm Cisco equipment uses the following feature restrictions: -Minimum signal duration is 100 milliseconds (ms). If a request is received with a duration less than 100ms, the minimum duration of 100 ms is used by default. -Maximum signal duration is 5000 ms. If a request is received with a duration longer than 5000 ms, the maximum duration of 5000 ms is used by default. -If no duration parameter is included in a request, the gateway defaults to a signal duration of 250 ms. Can this be what is causing my problem ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: meetme - conf-invalid
In article [EMAIL PROTECTED], Kevin Bockman [EMAIL PROTECTED] wrote: Yes, meetme requires a clock source. You could try ztdummy. I tried using an FXO card as a clock source and observed that SIP calls connected to the conference seemed to get out of sync. Basically, after perhaps 20 minutes or so in conference there was a 2 - 3 second delay between the time that one party spoke and the other party heard what was said. I have not tried ztdummy myself. Has anybody else seen this? According to http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy , ztrtc is supposted to fix the MeetMe delay. It compiles by default on Linux 2.6 with newer versions of -HEAD. Only if you remove the #if 0 from around #define USE_RTC Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: meetme - conf-invalid
In article [EMAIL PROTECTED], qrss [EMAIL PROTECTED] wrote: Yes, meetme requires a clock source. You could try ztdummy. I tried using an FXO card as a clock source and observed that SIP calls connected to the conference seemed to get out of sync. Basically, after perhaps 20 minutes or so in conference there was a 2 - 3 second delay between the time that one party spoke and the other party heard what was said. I have not tried ztdummy myself. Has anybody else seen this? Yes. Try the patch at http://bugs.digium.com/view.php?id=4252 to see whether it helps. Please post your results to that bug - thanks! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Junk at the beginning of frame
We upgraded to cvs-head a couple of days ago, and now get a whole slew of warnings in the error log: interface.c: Junk at the beginning of frame xx has anyone else seen this, or do I need to incur the wrath of developers and post this to the -dev list ;) Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Long time to detect hang-up
Title: Message Hi, Has anyone experienced the same problem. My telco provider is SBC. Regards,Stojan Sljivic -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDSSent: Tuesday, June 14, 2005 14:48To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Long time to detect hang-up Hi, I use Asterisk 1.0.5 and TDM04B. When an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk realize that. How can I shorten the time of hang-up detection? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Viva Madrid!
Agreed, I will post some pics early next week:) Wojtek - Original Message - From: Nicols Gudio [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 16, 2005 8:27 PM Subject: [Asterisk-Users] Viva Madrid! enough said -- Nicols Gudio Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ata186 IVR problem
Hi; Connected ata186 sip version 3.1.1 to IVR system...able to call / receive calls from ata but does not accept any dtmf When internal extension is dialed, it is not recognized and continues IVR musicDoes antbody has any idea how to make dtmf configuration from ata186 sip version?...it was used with sccp and cm without problem of dtmf Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mmkn olmayabilir. Mesaji alan kisi, mesajin gnderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkrler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Sipura 3000
[outgoing] ignorepat = 9 exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,3,Playback(nomoreline) exten = _9.,4,Hangup Chris - Original Message - From: Martin Roy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 16, 2005 11:41 PM Subject: [Asterisk-Users] Multiple Sipura 3000 If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for outgoing call. Can I create a group like I did in zaptel for Sipura 3000 device? Like if the FXO port of the first Sipura 3000 is busy it will switch to the second and if second is also busy then to the third one, and all the way until all the Sipura 3000 are in used before saying that there's no line left? The only configs I saw on the wiki were with 1 Sipura 3000 but I couldn't find anything on how to setup multiple Sipura 3000 devices in asterisk for outgoing calls. I would set it up the same way I have currently the zap channel configure so like this : [outgoing] ignorepat = 9 exten = _9.,1,Dial(Zap/g2/${EXTEN:1}) exten = _9.,2,Playback(nomoreline) exten = _9.,3,Hangup I tried this and it's working : [outgoing] ignorepat = 9 exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,2,Playback(nomoreline) exten = _9.,3,Hangup 10.0.1.111:5061 is the IP and SIP port of the Sipura 3000 device. So that would work great if I had only one Sipura but if I have multiple I would do it that way ? : [outgoing] ignorepat = 9 exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,6,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061) exten = _9.,7,Playback(nomoreline) exten = _9.,8,Hangup would that work? it's not quite the best thing to do as if I leave all the Sipura 3000 devices on DHCP if the IP ever change it will stop working and if one line is busy what will happen... Thanks Martin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial timeout when server down
Hello, When dialing somewhere and the other side is down, Asterisk waits until dial timeout before sending CHANUNAVAIL. I think that if after several seconds there are not any reply (I mean at the IP level) we could consider that the link is just down and handle the situation. Is it possible to configure Asterisk to have this behaviour? Many thanks. Yves. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?
Robert Rozman wrote: Is framing and coding (ami,ccs) right for Italy ? They are dummy settings with bristuff. The example config will surely do :) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comprehensive Asterisk Load Testing
The call generator is very not user friendly now and undocumented, i recommend not to use it and use some simple script somewhere for now. (you could find some scripts somewhere on astertest.com). I will put fixing that callgenerator on the (big) todo list. Zoa, Matt wrote: Has anyone gotten this tester to work? i can get it to log in and show me my call load.. but it doesn't seem to MAKE any calls. On 3/30/05, Kristian Kielhofner [EMAIL PROTECTED] wrote: Bicom Systems wrote: [EMAIL PROTECTED] wrote: Its a very very bad idea to do this on production boxes. Especially if you are trying to see how far you can go, and then you cross that tiny border :) Your production calls will not like an idle cpu% of 0% and a load of 500. I could not agree more with you hence my question :) However, the tests results produced on test boxes: How realistic it is? Does it really presents real life scenarios and results? Does it take in consideration different type of services (calls, IVR, queues) ? I am not trying to put down anyone or anything here, I am just curious. Ta Senad Senad, I have yet to take a real hard look or contact Zoa, but if all you are doing is calling an extension (very rapidly and many, many times) it really would not be very hard to test queues, music on hold, meetme, etc. I am downloading the callgenerator from astertest.com right now... The most realistic test is to (obviously) register as many phones as possible and hire hundreds of people to talk on them... :) -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial timeout when server down
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Yves Sent: Friday, June 17, 2005 2:09 AM To: Asterisk - Users Subject: [Asterisk-Users] Dial timeout when server down Hello, When dialing somewhere and the other side is down, Asterisk waits until dial timeout before sending CHANUNAVAIL. I think that if after several seconds there are not any reply (I mean at the IP level) we could consider that the link is just down and handle the situation. Is it possible to configure Asterisk to have this behaviour? If you're refering to an IAX channel then, yes, that's the concept behind the 'qualify=' option. However, there are known weaknesses including that the loss of a single ping makes the remote host appear to be down. There is a patch for head available at http://bugs.digium.com/view.php?id=4192 that attempts to make peer qualification more useful. Hope that helps. Kris Boutilier Information Services Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
Hi, I have the SuSe9.2 installed in a box with a QuadBri. I have followed all the instructions i have found and this is my best result... only one error compiling zaptel :( .. y have the kernel sources an already made the links to its drwxr-xr-x 8 root root 328 Jun 16 20:20 . drwxr-xr-x 12 root root 320 Jun 16 14:05 .. drwxr-xr-x 3 root root 136 Jun 16 20:17 asterisk drwxr-xr-x 22 root root 664 Jun 16 20:20 kernel-modules lrwxrwxrwx 1 root root 14 Jun 16 18:01 linux - linux-2.6.8-24 lrwxrwxrwx 1 root root 14 Jun 16 18:03 linux-2.6 - linux-2.6.8-24 drwxr-xr-x 21 root root 864 Jun 17 10:27 linux-2.6.8-24 drwxr-xr-x 3 root root 72 Oct 6 2004 linux-2.6.8-24-obj drwxr-xr-x 3 root root 72 Jun 16 17:28 linux-2.6.8-24.16-obj lrwxrwxrwx 1 root root 18 Jun 16 18:01 linux-obj - linux-2.6.8-24-obj drwxr-xr-x 7 root root 168 Jun 16 14:08 packages I made the make menuconfig and make dep in the kernel sources. but when i try to complile zaptel from the britstuff package this is the result: linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # make clean rm -f torisatool makefw tor2fw.h rm -f zttool ztspeed zttest ztmonitor rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core rm -rf .tmp_versions linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest You do not appear to have the kernel sources for your current kernel installed. make: *** [linux26] Error 1 linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # Why dont found the kernel sources? Help please Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
Americo 60+60 isn't a VoIP term directly but a generic one within the telephony industry if it were 60+30 it would mean the following You are billed for 60 second as soon as the call is answered, even if you only stay on the line for 7 seconds The +30 then referes to the onward billing cycle, so in this case you are billed in blocks of 30 seconds (ie if you call is 1 min 15 seconds you are billed for 1 min 30) You said that you are billed for a whole second minuite if you go over by even 1 second, so that would be a +60, and since its always a bigger or equal number first we are guessing that you are in 60+60 rate plan I think its more common in your part of the world for your carriers to bill 30+6. One in the replys suggested a very favorable rate of 6+6 The important thing to rember here is that you can't gaurentee enough return if you do a billing rate that is better than that of your carriers - it sounds to me like your offering your service on a 1+1 (ie true per second billing) rate - very honarable, but your carrier needs to offer the same. I hope that helps On 17/06/05, Americo Sanchez C. [EMAIL PROTECTED] wrote: From: Leon Sun [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Bill seconds Date: Thu, 16 Jun 2005 10:56:23 -0700 The easiest way is to change another vendor asap. Do you mean to change to another telecom? In my country there is a telephone monopoly :( Telefonica del Peru) It is ridiculous that your carrier still uses 60+60 now(30+6 is an asset). 2 seconds doesn't matter and billing unit does. Sorry I am not an expert in VoIP, What is the meaning of 60+60? Leon Sun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: June 15, 2005 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Consigue aqu las mejores y mas recientes ofertas de trabajo en Amrica Latina y USA: http://latam.msn.com/empleos/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Miax: Digital voice channel when connecting to asterisk
Hi, I've bought a Siemens GSM-modem based on the Siemens TC35-module. I studied the operation manual of the modem and found, that for transferring voice via the RS232 wire, the module supports RS232-mulitplexing and wires the voice data on a separate channel (whatever this means on RS232?). Now I wonder, whether that feature is supported by miax. All what I read about, was transfering GSM voice data via bluetoth from a cell phone to miax. Does anyone succeeded in connecting a GSM-modem via miax to asterisk and transfering the GSM voice data via the RS232 cable? Thanks for any hints! Roger. P.S. Somewhere I read the advice, that I should connect the (analogue) audio connector to the PC's soundcard, which is supported by miax. But transferring the GSM voice data analogious and digitizing again afterwords is not, what I really am looking for. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Commands D Option Question
When using the dial command and the D option to send DTMF digits when the channel is answered, is there a way to allow for some dead air, and then send more DTMF digits? I would like to automate a call, and it requires entry of a few short dtmf digits all a couple seconds apart from each other. Might look at the w within the dial string, which adds delay for each w appearing in the string. Something like: exten = _1NX,3,Dial(IAX2/xyz-itsp/9ww${EXTEN}) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Includes include the includes?
First, let me apologize for the multiple posts - my procmail recipe had a bug that hid most mail form the list for a day. The inheritance of includes creates a problem for me. I want to group the extensions, not put them all in default to control access to features. So [office] extensions should have the include = longdistance but [building1] should not. However, how can [building1] then dial office? Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie Sent: Wednesday, June 15, 2005 9:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Includes include the includes? Yes it does. You want something like this... [office] include = default include = local include = international [building1] include = default include = local [default] exten = 700,1,Dial(SIP/${EXTEN}) exten = 100,1,Dial(SIP/${EXTEN}) exten = 200,2,Dial(SIP/${EXTEN}) ;and so on for your other extensions ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call group channel limits
I have a question that I've so far been unable to find the answer to: Using an E1 interface for my PSTN connection I want to setup 5 SIP phones in a call group (with a unique number for inbound calls) but only allow the call group to receive a maximum of 3 calls at any one time. Does Asterisk have this functionality and if so, could someone point me to some examples?? Regs. Iain. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Sipura 3000
If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for outgoing call. Can I create a group like I did in zaptel for Sipura 3000 device? Like if the FXO port of the first Sipura 3000 is busy it will switch to the second and if second is also busy then to the third one, and all the way until all the Sipura 3000 are in used before saying that there's no line left? The only configs I saw on the wiki were with 1 Sipura 3000 but I couldn't find anything on how to setup multiple Sipura 3000 devices in asterisk for outgoing calls. If I understood what you're trying to accomplish, try something like this. In sip.conf, define each spa3k something like this: [3021] ; PSTN side of SPA3000 type=friend host=dynamic username=3021 secret=myspa1 context=from-sip canreinvite=no group=17 pickupgroup=2 deny=0.0.0.0/0.0.0.0 permit=216.21.194.0/255.255.255.0 and be sure to include group=17 in each spa3k definition. Then in extensions.conf, use a dial statement like this: exten = _9.,1,Dial(SIP/g17/${EXTEN:1} Pick whatever group number you want instead of =17 in the above. If I recall correctly, you can have up to 32 groups (or something like that). When the spa3k first hit the market, someone recommended using port 5060 and 5061 in the spa definitions. I have never had to do that with any spa3k. Rather, I leave both the fxs and fxo definitions in the spa3k default to 5060 and use different userid secrets for the fxs and fxo definitions. The above definition for x3021 is the actual one in use right now, which functions correctly. I've added the group=17 in the above as an example; I don't actually use that right now (for different reasons). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dell PowerEdge + TDM
Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog modems behind an Asterisk server?
Hello, we would like to hook up analog modems behind an Asterisk server, and we're very interested in the experiences that others have made when attempting that. We assume that there are no inherent problems with modems in respect to the Asterisk software, but it appears that the FXO/FXS hardware restricts this kind of a setup to lower data transmission rates, is this correct? Currently, we only transmit at 1200bps, is this rate problematic with Digium cards? Up to what data transmission rate are Digium cards known to work reliable? We do not think we'll ever go beyond 9600bps, can we do this with a let's say TDM400P? Will future Digium hardware improve the situation or will this stay the same in the future? How is hardware from other vendors performing when using analog modems? Thanks for any information on this, Christian Schnell. REKOBA GmbH Berlin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog modems behind an Asterisk server?
Hi, -Original Message- Currently, we only transmit at 1200bps, is this rate problematic with Digium cards? Up to what data transmission rate are Digium cards known to work reliable? We do not think we'll ever go beyond 9600bps, can we do this with a let's say TDM400P? On a pure TDM path this should be fine. In fact I think you should not have any real limitation if you set everything correctly. Using VoIP in parts of the link does limit the connection, although we have seen 14k4 connections run stable for a long time. YMMV. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog modems behind an Asterisk server?
Hello, we would like to hook up analog modems behind an Asterisk server, and we're very interested in the experiences that others have made when attempting that. We assume that there are no inherent problems with modems in respect to the Asterisk software, but it appears that the FXO/FXS hardware restricts this kind of a setup to lower data transmission rates, is this correct? Currently, we only transmit at 1200bps, is this rate problematic with Digium cards? Up to what data transmission rate are Digium cards known to work reliable? We do not think we'll ever go beyond 9600bps, can we do this with a let's say TDM400P? Will future Digium hardware improve the situation or will this stay the same in the future? How is hardware from other vendors performing when using analog modems? The success rate of moving modem data through a digium analog TDM card (fxo fxs ports) varies and appears to be somewhat related to the exact motherboard in use. The card very very frequently has an issue with missed data across the pci bus (card to motherboard). The missed data negatively impacts any modem call regardless of whether its a fax machine or pc modem. You are likely to have less then a 50% chance of making work correctly. Before a pile of people jump in to say it works for me, keep in mind that various types of modems use different modulation schemes and some are more sensitive than others to distorted audio (missed data). In very general terms, the higher the modem speed the more likely it will be negatively impacted by the distortion (missed data). If you're not familiar with modem technology, I might add there are two primary items that are directly related to the modem's audio across analog lines. The baud rate of analog signal on the wire and the bit rate of encoded digital data. You might have a current modem that allows you to change the bit rate (digital side), but on most modems you have no control over the analog baud rate (or modulation scheme). So, changing the modem's bit rate won't impact how well the modem actually works through the TDM card. Some people have reported that point of sale and credit card authorization boxes have worked via the TDM card. However, the modem's used in that equipment typically are very slow speed modems that were intended to function in any business environment including those with noisy telephone lines. Those have a higher possibility of success, but should not be interpreted as being the same as a modem used with PC's, etc. Bottom line... you will have far less then a 50% chance of making any PC modem work at acceptable speed through a TDM card. The latest code for the Sipura boxes (spa3k) appear to have addressed modem signals (fxs to fxo). I just upgraded two spa3k's to that latest firmware, but have not attempted to use any modem through it. Might check to see if anyone else on the list have tried it. The firmware was just released in the last day or two, so it might take a little while for folks to try it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents/Queues Contexts
Is there a way to define multiple contexts for agents/queues such that in a multi-tenant environment, there could be two different, say, Agents 1000? I'm setting up a multi-tenant configuration and I'm giving each tenant a web-based interface to define their own agents and I wouldn't want to restrict one tenant from choosing an agent because another tenant (which the tenant has no idea about) has already chosen that agent id. Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk Google API applications - $4500 bounties available
Btw here is an article on google maps that I wrote about the other day. http://www.smh.com.au/news/Technology/Map-hacks-make-data-come-alive/2005/06/16/1118869033845.html This is one of the best examples for www.craigslist.com I have ever seen http://housingmaps.com/ Cheers, Dean From: Dean Collins Sent: Wednesday, 1 June 2005 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Asterisk Google API applications - $4500 bounties available In conjunction with my last post on Tellme I want to write another suggestion for an application I had. I dont know if you guys have come across Google Gas http://www.ahding.com/cheapgas But basically it is an application that this guy has developed using the Google API to search an online database on gas prices in your area. One of my strong beliefs about how Asterisk is going to leave the Commercial IP-PBX vendors behind is by leveraging the open source community to write voice driven applications for Asterisk. The weather app written for [EMAIL PROTECTED] is great example. (http://sourceforge.net/forum/message.php?msg_id=3004652 the WAF on this was worth setting up asterisk alone, she checks this every morning for NY weather). I was also hoping that the www.tellme.com and www.studio.tellme.com tools would also stimulate this area. People should also check out www.angel.com for other ideas on best of breed speech applications. The suggestion I would like to make is that someone use the Google api to write code for a directions application. You could use Tellme to deliver the current address and the destination address into the Google API and then use text to speech to read back the directions. With enough finessing this could compete with any of the current commercial direction solutions that are out there and because its asterisk your cost base could be extremely minimal. Hell you might even get paid for it http://code.google.com/summerofcode.html Just a suggestion, any thoughts? Are there any other speech driven apps being used today? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge + TDM
Here we have PowerEdge 2850's doing the donky work with a Wildcard TE405P in each. I have seen no operational issues at all with the system or the cards. We are running CentOS 3 as the operating system and the stable version of asterisk The only niggle is that when the cards are modprobed on start up they sometimes 2 in a 100 give an NMI message, causing an error code on the servers little window, its not affected the stability at all, and its on my list of things to do to find out what causes it! The systems generally have around 400 - 500 SIP extensions comming off the back, running around a dual xeon 3Ghz and 3Gb of ram (no transcoding all G711.u) - we are very happy! David On 17/06/05, David Hajek [EMAIL PROTECTED] wrote: Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Last App-Fax Source
Can show me some link for the Last FaxApp sources (working with last spandsp - i think pre18) This link is not working. ftp://ftp.soft-switch.org/pub/spandsp. ftp://ftp.soft-switch.org/pub/spandsp The Domain soft-switch.org ftp://ftp.soft-switch.org/pub/spandsp is not resovable. And what is the version of libtiff I must install? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge + TDM
Thanks for the reply. I'm more interested in lower series then 2850, like PE SC1420, PE 800. I don't need so much power. ;-) -David Here we have PowerEdge 2850's doing the donky work with a Wildcard TE405P in each. I have seen no operational issues at all with the system or the cards. We are running CentOS 3 as the operating system and the stable version of asterisk The only niggle is that when the cards are modprobed on start up they sometimes 2 in a 100 give an NMI message, causing an error code on the servers little window, its not affected the stability at all, and its on my list of things to do to find out what causes it! The systems generally have around 400 - 500 SIP extensions comming off the back, running around a dual xeon 3Ghz and 3Gb of ram (no transcoding all G711.u) - we are very happy! David On 17/06/05, David Hajek [EMAIL PROTECTED] wrote: Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream
But when BT-100 calls 7960 the following is happening: -- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack -- Called 1707 -- SIP/1707-e96a is ringing -- SIP/1707-e96a answered SIP/3710-8f2b -- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a May 4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. sipsrv1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 192.168.128.171 170702fff7f7169 00102/0 ulawTx: ACK 67.126.23.2513710b5d3f977ea1 00101/52181 g729Rx: ACK When this bug is gonna be fixed? Change the codec order in the phone configuration and place g729 higher it is not asterisk doing this ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk box as a billing machine in a PSTN network
Hi, Is it possible to use an Asterisk box as billing gateway in a PSTN network? (Asterisk box somewhat connected to PSTN switch)? In case the answer to the above question is yes, how to proceed? Thanks, Simon Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Tlchargez le ici ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Sipura 3000
I saw this in the list awhile back, it helped me setup my sipura 3000s to act as trunks Setup the PSTN side of the Sipura 3000 as a trunk within Asterisk In AMP add an extension (e.g. 200) to correspond to Line 1 on the SPA, ensure that port is 5060 and context is from-internal. It should be named SatelliteOut. Disable voicemail and directory. Add a second extension (e.g. 201) for PSTN Line on SPA, ensure that port is 5061 and set context to from-pstn. It should be named SatelliteIn. Disable voicemail and directory. In Trunks add a Sip trunk and copy the Outgoing block as follows (just leave Incoming as it is - do not delete the any defaults, but you do not need to change them either).: Trunk name SatelliteOut context=from-pstn fromuser=201 (or whatever extension you used) host=IP address of you SPA (needs to be fixed IP) port=5061 secret=your password type=peer username=201 (or whatever extension you used) Inbound User context SatelliteIn Leave defaults in Inbound box and leave Register String blank. In DID Routes, add DID with a unique string (I used S followed by the PSTN number that the SPA is attached to - e.g. S888777 Set an outbound route using the new SatelliteOut trunk. On the SPA 3000: Do the following configuration in admin login, advanced mode: In Line 1, make sure SIP port is 5060, proxy points to your * Box, NO outbound proxy. Fill out subscriber info with settings above e.g. User ID = 200 Password =1234 Display Name = SatelliteIn In PSTN Line, ensure SIP Port = 5061 proxy = Asterisk Box IP, NO outbound proxy. Fill out subscriber info with Display Name = SatelliteOut User ID = 201 Password =1234 It is vital that you Set Dial Plan 8 to (S0:S888777) (for the string you used for the DID route in Asterisk). Ensure that both VoIP-To-PSTN Gateway Enable and PSTN-To-VoIP Gateway Enable are set to yes. Set PSTN Caller Default DP to 8. If you want incoming calls to all be sent to * then set PSTN Ring Thru Line 1 to no. Set PSTN Answer Delay to the number of seconds that you want the phone to ring for before sending it to your * box. Set it to 1. Leave other settings on the SPA at factory defaults until you really know what you're doing and want to fine-tune things. Lastly, make sure you plug into the line jack into the SPA and not the jack marked phone! I know this seems obvious, but I've missed this simple step before! The only kink with inbound using the settings posted is that you can't have it ring to a phone plugged into the Sipura's phone port. You can still call out, and the system will still pick up the call if you have auto attendant recieve the calls. But, if you set the inbound calls to ring extension 200, your calls will just go directly to voicemail. That aside, you can have any other phone on the system ring for inbound calls directly, or set a ring group. On 6/17/05, Rich Adamson [EMAIL PROTECTED] wrote: If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for outgoing call. Can I create a group like I did in zaptel for Sipura 3000 device? Like if the FXO port of the first Sipura 3000 is busy it will switch to the second and if second is also busy then to the third one, and all the way until all the Sipura 3000 are in used before saying that there's no line left? The only configs I saw on the wiki were with 1 Sipura 3000 but I couldn't find anything on how to setup multiple Sipura 3000 devices in asterisk for outgoing calls. If I understood what you're trying to accomplish, try something like this. In sip.conf, define each spa3k something like this: [3021] ; PSTN side of SPA3000 type=friend host=dynamic username=3021 secret=myspa1 context=from-sip canreinvite=no group=17 pickupgroup=2 deny=0.0.0.0/0.0.0.0 permit=216.21.194.0/255.255.255.0 and be sure to include group=17 in each spa3k definition. Then in extensions.conf, use a dial statement like this: exten = _9.,1,Dial(SIP/g17/${EXTEN:1} Pick whatever group number you want instead of =17 in the above. If I recall correctly, you can have up to 32 groups (or something like that). When the spa3k first hit the market, someone recommended using port 5060 and 5061 in the spa definitions. I have never had to do that with any spa3k. Rather, I leave both the fxs and fxo definitions in the spa3k default to 5060 and use different userid secrets for the fxs and fxo definitions. The above definition for x3021 is the actual one in use right now, which functions correctly. I've added the group=17 in the above as an example; I don't actually use that right now (for different reasons).
[Asterisk-Users] auto-dial dial status
I'm using autodial in conjuction with TxFax to send faxes on demand. An home made application generates the call file and puts it in the outgoing spool, the file is like this: Channel:Zap/g1/1232314324 MaxRetries:0 RetryTime:60 WaitTime:20 Context:faxout Extension:s SetVar:FAX_FILE=/shared/awfax/test.tif the extension called is this: [faxout] exten = s,1,TxFax(/shared/awfax/test.tif|caller) exten = s,2,Hangup My problem is that if asterisk can't connect to the called end then it doesn't go to the extension, so i am unable to report the error if the called end does not respond, does not exist or refuse the call. Is there some trick (or an elegant solution as well) to solve this problem? ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ringing tone on outgoing SIP trunk
Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is making progress and puts through the incoming RTP I suppose. With an external anonymous ATA from the VOIP provider all works normally. Here is the debug: -- Accepted AUTHENTICATED TBD call from 192.168.19.130 -- Accepting DIAL from 192.168.19.130, formats = 0x4 -- Executing Macro(IAX2/[EMAIL PROTECTED]/2;, dialout-trunk|2|0639006374|) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(IAX2/[EMAIL PROTECTED]/2;, record-enable|200|OUT) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 1?5:8) in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget(IAX2/[EMAIL PROTECTED]/2;, RecEnable=RECORD-OUT/200) in new stack -- DBget: varname=RecEnable, family=RECORD-OUT, key=200 -- DBget: Value not found in database. -- Executing SetVar(IAX2/[EMAIL PROTECTED]/2;, CALLFILENAME=OUT200-20050616-180439-1118959479.2) in new stack -- Executing Goto(IAX2/[EMAIL PROTECTED]/2;, s|14) in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 0?15:99) in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp(IAX2/[EMAIL PROTECTED]/2;, NO RECORDING NEEDED) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 1?7) in new stack -- Goto (macro-dialout-trunk,s,7) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 0?9) in new stack -- Executing SetCallerID(IAX2/[EMAIL PROTECTED]/2;, 0872596100) in new stack -- Executing SetGroup(IAX2/[EMAIL PROTECTED]/2;, OUT_2) in new stack -- Executing CheckGroup(IAX2/[EMAIL PROTECTED]/2;, ) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/2;, DIAL_NUMBER=0639006374) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/2;, DIAL_TRUNK=2) in new stack -- Executing AGI(IAX2/[EMAIL PROTECTED]/2;, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(IAX2/[EMAIL PROTECTED]/2;, OUTNUM=0639006374) in new stack -- Executing Cut(IAX2/[EMAIL PROTECTED]/2;, custom=OUT_2|:|1) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/2;, 0?19) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]/2;, SIP/micso/0639006374) in new stack -- Called micso/0639006374 -- SIP/micso-0626 is making progress passing it to IAX2/[EMAIL PROTECTED]/2 == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2' in macro 'dialout-trunk' == Spawn extension (from-internal, 90639006374, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2' -- Executing Macro(IAX2/[EMAIL PROTECTED]/2;, hangupcall) in new stack -- Executing ResetCDR(IAX2/[EMAIL PROTECTED]/2;, w) in new stack -- Executing NoCDR(IAX2/[EMAIL PROTECTED]/2;, ) in new stack -- Executing Wait(IAX2/[EMAIL PROTECTED]/2;, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2' -- Hungup 'IAX2/[EMAIL PROTECTED]/2' What can I do? Do you need somethign else? Thanks! Norm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell PowerEdge + TDM
I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to configure it using [EMAIL PROTECTED] scripts and did not work, so I went the long way and configure with zaptel's instructions and voila! It works like a charm. Oswaldo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Friday, June 17, 2005 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dell PowerEdge + TDM Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Includes include the includes?
If you set up only internal extesions in the default context, then include default in [building1] and [office] all of those extensions can call internally. I set up several standard features into the default context which everyone can access. If you want to control feature access, say, for example a line that reads the time (let's just say), put that in a different context. [office] include = default include = local include = international include = timeline [building1] include = default include = local [default] exten = 700,1,Dial(SIP/${EXTEN}) exten = 100,1,Dial(SIP/${EXTEN}) exten = 200,2,Dial(SIP/${EXTEN}) ;and so on for your other extensions [timeline] exten = something Or, if you wanted to control access to who could call which internal extension, then you break out default into groups of their own. [office] include = office-ext include = local include = international include = timeline [building1] include = building1-ext include = office-ext include = local [office-ext] exten = 700,1,Dial(SIP/${EXTEN}) [building1-ext] exten = 100,1,Dial(SIP/${EXTEN}) In that example, office can call other office numbers, make local calls, make international calls, and access the timeline feature. Building 1 can access building1 extensions, office extensions, and make local calls. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Mason (Lists) Sent: Friday, June 17, 2005 5:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Includes include the includes? First, let me apologize for the multiple posts - my procmail recipe had a bug that hid most mail form the list for a day. The inheritance of includes creates a problem for me. I want to group the extensions, not put them all in default to control access to features. So [office] extensions should have the include = longdistance but [building1] should not. However, how can [building1] then dial office? Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tarpo, Louie Sent: Wednesday, June 15, 2005 9:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Includes include the includes? Yes it does. You want something like this... [office] include = default include = local include = international [building1] include = default include = local [default] exten = 700,1,Dial(SIP/${EXTEN}) exten = 100,1,Dial(SIP/${EXTEN}) exten = 200,2,Dial(SIP/${EXTEN}) ;and so on for your other extensions ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2nd Dialtone after answer
Hi I am trying to achive this for a specific need of a customer. He has a DID pointed to an Asterisk server, I need to provide him dialtone when the calls hits the server. How can I achieve this? Let's say something like this: Exten = s,1,Answer Exten = s,2, Provide Dial tone Exten = s,3, Dial the number the person will enter after receiving the dial tone Exten = s,4,Hangup Any ideas? Thanks very much Oswaldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calculating the lenght of time in a call queue?
Dear All, I'm running version 0.7.1 of Asterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2nd Dialtone after answer
Check out DISA. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Oswaldo Arratia |Sent: Friday, June 17, 2005 7:51 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] 2nd Dialtone after answer | | |Hi |I am trying to achive this for a specific need of a customer. | |He has a DID pointed to an Asterisk server, I need to provide |him dialtone when the calls hits the server. How can I achieve this? | |Let's say something like this: | |Exten = s,1,Answer |Exten = s,2, Provide Dial tone |Exten = s,3, Dial the number the person will enter after |receiving the dial tone Exten = s,4,Hangup | |Any ideas? | |Thanks very much | |Oswaldo | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Programinng Aplication with Music on Hold
Hi, I am programing one aplications to hear my e-mail on PBX asterisk. I want to have music on hold when the my aplications get the e-mail the mail server real. I am doing un IVR to this aplicationes. How I can do that? When is the routines on the source asterisk to music on hold? What files I can see, for one example? Thank, Raul Pineda ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, June 15, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] WiFi IP Phones Guys. I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. You are referring to (in the US anyway) certification as intrinsically safe. I don't know either way about phones listed as such, but with the right terminology you might have better liuck searching. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2nd Dialtone after answer
it sounds like you need to investigate the application called DISA.. it might be what you are looking for if not, what dial tone is your client expecting? (internal, external - other??) David On 17/06/05, Oswaldo Arratia [EMAIL PROTECTED] wrote: Hi I am trying to achive this for a specific need of a customer. He has a DID pointed to an Asterisk server, I need to provide him dialtone when the calls hits the server. How can I achieve this? Let's say something like this: Exten = s,1,Answer Exten = s,2, Provide Dial tone Exten = s,3, Dial the number the person will enter after receiving the dial tone Exten = s,4,Hangup Any ideas? Thanks very much Oswaldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating the lenght of time in a call queue?
Time waiting for an agent is one of the fields recorded in the queue_log see the following http://voip-info.org/tiki-index.php?page=Asterisk%20log%20queue_log Regards, Mac. -Original Message- From: Shad Mortazavi [mailto:[EMAIL PROTECTED] Sent: 17 June 2005 15:54 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Calculating the lenght of time in a call queue? Dear All, I'm running version 0.7.1 of Asterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, June 16, 2005 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] WiFi IP Phones Ahmm Andrew, are you sure they are steel? It's been a long time since I did any work in this space but we used to install them in plastic not metal.plastic works better with the radio waves. IS does not necessarily mean steel. My Motorola alpha pager, and my Motorola XTS3000 radio are both plastic and IS listed. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Tuesday, June 14, 2005 3:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Anyone paying over $450 for a T1 is being ripped off... If you are in VA,MD,DC,PA,DE,NJ you can get an integrated VoIP T1 for $300 - $400 and a flat internet t1 for about $400. The integrated VoIP T1 is great because it's handed off as an ethernet - no need for a csu/dsu Ummm...no. Maybe if you are in or very near a city you can, but not everywhere. You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400 and I'll give you the amount I save over the next quarter. NPA-NXX is 215-862. Good luck. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFC/R2
The libsupertone library installation happened because the libxml2 library was not installed. After it was installed the problem was solved. Now I am experiencing the problem following: Do you have any tip??? Thanks. [chan_unicall.so] = (Unified call processing (UniCall)) == Parsing '/etc/asterisk/unicall.conf': Found Loading protocol mfcr2 Country 5 -- Registered channel 9, mfcr2 signalling Country 5 -- Registered channel 10, mfcr2 signalling Country 5 -- Registered channel 11, mfcr2 signalling Country 5 -- Registered channel 12, mfcr2 signalling Country 5 -- Registered channel 13, mfcr2 signalling Country 5 -- Registered channel 14, mfcr2 signalling Country 5 -- Registered channel 15, mfcr2 signalling Country 5 -- Registered channel 16, mfcr2 signalling Country 5 -- Registered channel 17, mfcr2 signalling Country 5 -- Registered channel 18, mfcr2 signalling Country 5 -- Registered channel 19, mfcr2 signalling Country 5 -- Registered channel 20, mfcr2 signalling Country 5 -- Registered channel 21, mfcr2 signalling Country 5 -- Registered channel 22, mfcr2 signalling Country 5 -- Registered channel 23, mfcr2 signalling Country 5 -- Registered channel 25, mfcr2 signalling Country 5 -- Registered channel 26, mfcr2 signalling Country 5 -- Registered channel 27, mfcr2 signalling Country 5 -- Registered channel 28, mfcr2 signalling Country 5 -- Registered channel 29, mfcr2 signalling Country 5 -- Registered channel 30, mfcr2 signalling Country 5 -- Registered channel 31, mfcr2 signalling Country 5 -- Registered channel 32, mfcr2 signalling Country 5 -- Registered channel 33, mfcr2 signalling Country 5 -- Registered channel 34, mfcr2 signalling Country 5 -- Registered channel 35, mfcr2 signalling Country 5 -- Registered channel 36, mfcr2 signalling Country 5 -- Registered channel 37, mfcr2 signalling Country 5 -- Registered channel 38, mfcr2 signalling Country 5 -- Registered channel 39, mfcr2 signalling Country 5 -- Registered channel 40, mfcr2 signalling Country 5 -- Registered channel 41, mfcr2 signalling Country 5 -- Registered channel 42, mfcr2 signalling Country 5 -- Registered channel 43, mfcr2 signalling Country 5 -- Registered channel 44, mfcr2 signalling Country 5 -- Registered channel 45, mfcr2 signalling Country 5 -- Registered channel 46, mfcr2 signalling Country 5 -- Registered channel 47, mfcr2 signalling Country 5 -- Registered channel 48, mfcr2 signalling Country 5 -- Registered channel 49, mfcr2 signalling Country 5 -- Registered channel 50, mfcr2 signalling Country 5 -- Registered channel 51, mfcr2 signalling Country 5 -- Registered channel 52, mfcr2 signalling Country 5 -- Registered channel 53, mfcr2 signalling Country 5 -- Registered channel 54, mfcr2 signalling Country 5 -- Registered channel 56, mfcr2 signalling Country 5 -- Registered channel 57, mfcr2 signalling Country 5 -- Registered channel 58, mfcr2 signalling Country 5 -- Registered channel 59, mfcr2 signalling Country 5 -- Registered channel 60, mfcr2 signalling Country 5 -- Registered channel 61, mfcr2 signalling Country 5 -- Registered channel 62, mfcr2 signalling Country 5 -- Registered channel 63, mfcr2 signalling Country 5 -- Registered channel 64, mfcr2 signalling Country 5 -- Registered channel 65, mfcr2 signalling Country 5 -- Registered channel 66, mfcr2 signalling Country 5 -- Registered channel 67, mfcr2 signalling Country 5 -- Registered channel 68, mfcr2 signalling Country 5 -- Registered channel 69, mfcr2 signalling Country 5 -- Registered channel 70, mfcr2 signalling Parsing tone set Hit dial-tone Step - Frequency=350.00+440.00 [1.00%]Level=-13.00+-13.00 Recognition length=0.30 [10.00%] Detector element350440300 0 Hit dial-tone Step - Cycles=3 Step - Frequency=350.00+440.00 [1.00%]Level=-13.00+-13.00Length=0.10 [10.00%] Detector element350440 60140 Step - Length=0.10 [10.00%] Detector element 0 0 60140 Step - Frequency=350.00+440.00 [1.00%]Level=-13.00+-13.00 Recognition length=0.30 [10.00%] Detector element350440300 0 Hit ringback-tone Step - Cycles=0 Step - Frequency=440.00+480.00 [1.00%]Level=-13.00+-13.00Length=0.40 [10.00%] Detector element440480330470 Step - Length=0.20 [10.00%] Detector element 0 0150250 Step - Frequency=440.00+480.00 [1.00%]Level=-13.00+-13.00Length=0.40 [10.00%] Detector element440480330470 Step - Length=3.00 [10.00%] Recognition length=0.60 [10.00%] Detector element
RE: [Asterisk-Users] WiFi IP Phones
Given that they are radio transmitters, there is always the risk that they can cause a spark and ignite something. Additionally, reports have happened of the battery itself getting shorted when removed and causing everything from bullets to other explosive situations to occur. When you short the metal contacts on the battery it gets very hot, that heat can cause things to ignite (gasoline for example needs about 500 degrees farenheight to ignite). Cell phones are normally upto 600mW PEP (max FCC allowed for a handheld). Wifi devices are normally upto 200mW PEP. Amplifiers can change this. ETSI in europe limits to 100mW PEP and in Japan wifi is limited to 10mW or something silly. Thus its less of a concern in other regions. If you put a little bit of metal in the microwave you will see sparks on the metal. This is because there is a difference in RF potential across two points, and an arcing occurs. Granted most microwaves are 600-1000W PEP, or 1000+ times the power, but the same type of situation can occur if conditions are right. In short there is no way to completly reduce the chance of explosions of certain substances, to get a cell phone far enough away to mitigate that danger is a matter of inches (1 inch == 2.54 cm), a wifi device, having less power, is an even shorter distance. If you are very near dangerous substances that could be set off this way you should (hopefully anyway) be trained in proper procedure there. There is more risk (I think anyway) of filling a plastic gas can inside a pickup bed with a plastic liner (plastic on plastic can create a static discharge). Cigarettes often dont get hot enough to ignite gasoline (outside movies) because only when inhaling do they near hot enough, just tossing one onto gas its normally 50 degrees below the flash point. Remember liquids dont burn, only gasses do in normal physics anyway (special conditions can occur with extreme temperatures and pressures). Gas station tanks are grounded if metal, the pumps certainly are. This further mitigates risk. AFAIK there arent regulations that require them to prevent explosions, any regulations like that would be on the devices that contain or transport such materials that are likely to explode. Because people transmit a lot of power on mobile radios, those working with detonators at construction sites often are required to put out signs saying 'dont transmit explosive danger' because you can cause a false fire signal to be sent to the detonator if you kick out enough power, but even those devices are typically shielded to minimize this risk. On Fri, 2005-06-17 at 10:49 -0400, Daryl G. Jurbala wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, June 15, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] WiFi IP Phones Guys. I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. You are referring to (in the US anyway) certification as intrinsically safe. I don't know either way about phones listed as such, but with the right terminology you might have better liuck searching. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Presence and IM?
We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
Manuel Casal ha scritto: I made the make menuconfig and make dep in the kernel sources. i do not remember well how i solved that problem but i'm sure that make dep will issue you a warning and stop. run make to start the kernel build process and then stop it after few seconds. it will create the necessary symlinks in the kernel tree. maybe there's a more elegant solution but this should work. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2nd Dialtone after answer
Hi all, DID is Direct Inward Dial where the last 3 or 4 digits of the dialed number are passed through and are used/translated to call a specific extension. (See Centrex) DISA is Direct System Access where incoming line(s)are auto-answered and receive internal dial tone, the caller then has access to the facilities of the system.(including calling an extension.) I hope this clears things up TTFN Henry Chris Coulthurst wrote: Check out DISA. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Oswaldo Arratia |Sent: Friday, June 17, 2005 7:51 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] 2nd Dialtone after answer | | |Hi |I am trying to achive this for a specific need of a customer. | |He has a DID pointed to an Asterisk server, I need to provide |him dialtone when the calls hits the server. How can I achieve this? | |Let's say something like this: | |Exten = s,1,Answer |Exten = s,2, Provide Dial tone |Exten = s,3, Dial the number the person will enter after |receiving the dial tone Exten = s,4,Hangup | |Any ideas? | |Thanks very much | |Oswaldo | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Presence and IM?
Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't switch span to E1-mode
Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16 bchan=17-31 [EMAIL PROTECTED] etc]# cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] signalling=pri_cpe immediate=no switchtype=EuroISDN pridialplan=unknown context=incoming usecallerid=yes group=1 channel = 1-23 channel = 17-31 [EMAIL PROTECTED] etc]# /etc/init.d/zaptel restart Unloading zaptel hardware drivers: wcusb wctdm wcfxo wcte11xp wct1xxp wct4xxp tor2. Removing zaptel module:[ OK ] Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules:Running ztcfg: ZT_CHANCONFIG failed on channel 25: No such device or address (6) [FAILED] [EMAIL PROTECTED] etc]# dmesg TE110P: Setting up global serial parameters for T1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 [EMAIL PROTECTED] etc]# ztcfg -v Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. ZT_CHANCONFIG failed on channel 25: No such device or address (6) Now I'll show how it work with 24 channels [EMAIL PROTECTED] etc]# dmesg TE110P: Setting up global serial parameters for T1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Registered tone zone 11 (Italy) TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) Registered tone zone 11 (Italy) TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) Registered tone zone 11 (Italy) usbcore: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) Registered tone zone 11 (Italy) TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) Registered tone zone 11 (Italy) And no errors. Help, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
cheapest in the world. Ha Ha Ha Ha How is your electricity sold? Hour, Watt, or by unit (KW/h)? And as for cell phones being cheap, you have a receiver pays setup! How good is that, then you have so many competing Telcos that sometimes you just cannot call the house across the street without tracversing 3 exchanges, and then you have so many area codes (each of which is so small) that almost all your calls are STD, then as for cheap, how is AUD$49.00/Month for all you can eat (that's about US$30/month) and all incoming calls are free, YES FREE! :) Really??? http://www.cucumber.com/fullinternational29.htm Look at the difference they (and everybody else) charges to call the cell network. Check your bill (your landline bill) next month and tell hoe much cell phone costs you. So much for the american(sic) always making things better. I knew that none americans might not see this unless pointed out to them :) The best phone and postal services by far in in Australia. There are no peers at all. Sorry, I just had to call that bluff T ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
On 6/16/05, Tony Hoyle [EMAIL PROTECTED] wrote: Terry H. Gilsenan wrote: And as for cell phones being cheap, you have a receiver pays setup! How good is that, then you have so many competing Telcos that sometimes you just I believe that's unique to the US, the idea of paying for actually receiving calls... don't know why they stand for it but I guess the masses don't know any better. Here in the UK mobile phones are only 15/month on the cheapest tariffs which equates to about $25.. $.30 a min to UK cell how is this cheap? http://www.cucumber.com/fullinternational29.htm This thread could easily become a 'my phone is cheaper than yours' argument :) I'm just waiting for someone from Japan to post... (they *are* the cheapest in the world I believe, by a long margin). As for Japan I don't know you might be right. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729
Title: Untitled Document Hi All, I have configured Line1 (2011)and Line2(2012)in SipuraSPA-2000 (latest Firmware)to use G729. In sip.conf I have set disallow=all, allow=g729 IfLine1 is in use by an agent, then Line2 won't work and viceversa (Inbound Calls Only).I have 40 license for G729. so there shouldn't be any issue with the license. I'm getting the following error msg: -- Called 2012 -- Got SIP response 488 "Not Acceptable Here" back from 192.168.10.103 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' status is 'NOANSWER' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' If I change 2012 to ULAW, it works fine. It seems that I can't have two lines configured as a G729. Do you guys have any idea why this happening? Regards, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tdm400p not working after cvs-head update
I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks, today i did a CVS update to the latest head files and the card is not working. Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) HELP!. thanks David Romero## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP transfer/REFER to voicemail problem
For anyone else who might run into this, I got around the transferring to voicemail problem by putting a canreinvite=no line into the section for each caller's SIP address in sip.conf. Not ideal, but it works. I also had to add a dtmfmode=inband for my Mediatrix 1204 addresses to be able to access the voicemail commands. -- I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't answer. Caller (A) hears the Asterisk voicemail prompts, but the voicemail application doesn't hear any audio or DTMF. Easy to duplicate: 1.) A - B (INVITE) 2.) B - C (REFER A to C) 3.) A - C More descriptive: 1.) Caller (A) calls intermediary (B). (B can be any SIP user agent) 2.) Intermediary (B) REFERs caller (A) to callee (C) 3.) C is either a SIP UA which times-out and Asterisk takes to Voicemail, or an extension tied to VoicemailMain. I've come across a thread saying that the Asterisk voicemail system only uses the GSM codec, but if this were the problem, then how can the caller (using mu-law) hear the voicemail prompts? Would Asterisk be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host=dynamic disallow=all ;allow=gsm allow=ulaw extensions.conf exerpt: exten = 6003,1,Dial(SIP/1003,15) exten = 6003,2,Voicemail(u1003) exten = 6003,102,Voicemail(b1003) exten = 6004,1,Dial(SIP/1004,5) exten = 6004,2,Voicemail(u1004) exten = 6004,102,Voicemail(b1004) exten = 2101,1,Dial(SIP/2101) exten = 8500,1,VoicemailMain exten = 8500,2,Hangup Asterisk (-dvvgc) with sip debug on (REFER-ing caller to VoicemailMain) : -- No username but # key pressed. Using CID '6003' -- Playing 'vm-password' (language 'en') Urgent handler -- Incorrect password '' for user '6003' (context = ,any) -- Playing 'vm-incorrect-mailbox' (language 'en') Urgent handler __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729
The Sipura SPA2000 only supports one G729 call at a time. Same with the Linksys PAP2. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Fri, 17 Jun 2005, David wrote: Hi All, I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest Firmware) to use G729. In sip.conf I have set disallow=all, allow=g729 If Line1 is in use by an agent, then Line2 won't work and vice versa (Inbound Calls Only). I have 40 license for G729. so there shouldn't be any issue with the license. I'm getting the following error msg: -- Called 2012 -- Got SIP response 488 Not Acceptable Here back from 192.168.10.103 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' status is 'NOANSWER' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' If I change 2012 to ULAW, it works fine. It seems that I can't have two lines configured as a G729. Do you guys have any idea why this happening? Regards, This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-06-17%5Cd81c0f432a8146fd9b6064a4b2fc65b8C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
If you tell an american 'this is it', s/he will think of ways to change it and make it another way, thats why we have a '96 telecommunications act, and why having cell phones in the states are the cheapest in the world. Oh yeah. Americans are always faster, better, bigger. We cheese eating surrending monkeys get the idea ;-) Can we go back to asterisk stuff now? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729
Title: Untitled Document Hi, The Sipura SPA-2000 can only support one G729 call Regards Erick - Original Message - From: David To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, June 17, 2005 11:33 AM Subject: [Asterisk-Users] G729 Hi All, I have configured Line1 (2011)and Line2(2012)in SipuraSPA-2000 (latest Firmware)to use G729. In sip.conf I have set disallow=all, allow=g729 IfLine1 is in use by an agent, then Line2 won't work and viceversa (Inbound Calls Only).I have 40 license for G729. so there shouldn't be any issue with the license. I'm getting the following error msg: -- Called 2012 -- Got SIP response 488 "Not Acceptable Here" back from 192.168.10.103 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' status is 'NOANSWER' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' If I change 2012 to ULAW, it works fine. It seems that I can't have two lines configured as a G729. Do you guys have any idea why this happening? Regards, ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
Marco Parmeggiani escribi: Manuel Casal ha scritto: I made the make menuconfig and make dep in the kernel sources. i do not remember well how i solved that problem but i'm sure that make dep will issue you a warning and stop. run make to start the kernel build process and then stop it after few seconds. it will create the necessary symlinks in the kernel tree. maybe there's a more elegant solution but this should work. ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I did that and this is the result... a new kind of error linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # make all cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 modules make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make: *** [linux26] Error 2 linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # Now what?:( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2
Manuel Casal ha scritto: make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp' make: *** [linux26] Error 2 linux:/usr/src/asterisk/bristuff-0.2.0-RC8g/zaptel-1.0.7 # Now what?:( i'm using a Debian. i'm missing those *-obj links in my /usr/src drwxr-xr-x 19 root root 4096 Jun 17 18:04 kernel-source-2.6.11 lrwxrwxrwx 1 root src18 May 24 14:36 linux - /usr/src/linux-2.6 lrwxrwxrwx 1 root src20 May 24 13:54 linux-2.6 - kernel-source-2.6.11 and it compiles fine. HTH ciao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txfax 18Kb file problem
Hi ALL. I have a problem with TxFax application. (RxFax is working properly) Txfax does not work when sending tiff files bigger than approx. 18Kb. If tiff file smaller than 18Kb everything is OK. Tested in LAN with Panasonic UF-E1 fax machine (tiff files was created by the rxfax from that machine) and SAP-3000. Also tested with some remote fax machines (connected to PSTN and via VoIP). Results are same. Small file in most cases is OK, bigger is FAILED. Environment: Fedora Core 2 and 3, spandsp 0.0.2pre18, libtiff 3.5.7-20.2 and 3.7.1-6, Asterisk 20050427CVS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't switch span to E1-mode
What card do you have ? Is there are jumper setup that you can specify E1 or T1 ? E1 cards a shipped set up as T1 by default regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge + TDM
Do you have analog TDM in it? -David Oswaldo Arratia wrote: I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to configure it using [EMAIL PROTECTED] scripts and did not work, so I went the long way and configure with zaptel's instructions and voila! It works like a charm. Oswaldo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Friday, June 17, 2005 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dell PowerEdge + TDM Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PIX Firewall Ports and Access-Lists
Hello, I am not too familiar with the settings in our PIX (learning though). Here is the only access-list setting that we have in place for Asterisk: access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060 In rtp.conf we are allowing ports 1 - 2. We are not using SIP Fixup in our PIX due to firmware version. How do I go about adding the ability for udp ports 1 - 2 to forward to our Asterisk server? We have intermittent audio issues on calls and I have narrowed it down (hopefully) to the PIX. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:Calculating the lenght of time in a call queue?
I don't get a queue_log file? At what stage was this introduced? Thanks Shad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phantom problem authenticating IAX2 with RSA
I'm getting exactly the same behavior as was posted about in http://lists.digium.com/pipermail/asterisk-users/2004-March/040380.html I've upgraded (both ends) to CVS stable (CVS-v1-0-06/17/05-13:15:49). Jun 17 13:46:17 NOTICE[15942]: chan_iax2.c:4053 authenticate: No way to send secret to peer 'a.b.c.d' (their methods: 4) Immediately after that, I'll see frames go by with Tx-Frame Retry[000] Subclass: NEW Rx-Frame Retry[ No] Subclass: AUTHREQ Tx-Frame Retry[000] Subclass: AUTHREP Rx-Frame Retry[ No] Subclass: ACCEPT that make it look very much like rsa authentication is being done, and the call is accepted. I noticed this while cleaning up my IAX config...moving away from type=friend entries to a type=user and a type=peer entry for each system I send/receive calls to/from. i.e. on the remote end, I have: [my.system.name] username=my.system.name type=user auth=rsa inkeys=my.system.name context=my.system.name-iax qualify=no disallow=all allow=g729 allow=gsm deny=0.0.0.0/0.0.0.0 permit=[IP of my.system.name] On the end I'm calling from: [remote.system.name] type=peer username=my.system.name auth=rsa outkey=my.system.name qualify=no disallow=all allow=g729 allow=gsm host=remote.system.name The test call is dialed as IAX2/remote.system.name/${EXTEN} Is there a problem with my config, or is this just an iax2 cosmetic bug? Each end does have appropriate rsa keys (readable by asterisk) in /var/lib/asterisk/keys. BTW, if I'm reading the docs correctly, there are multiple errors in the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20IAX%20authentication#comments where allow is incorrectly used [in the context of allowing an IP] where permit was meant. -- Jon Lewis | I route Senior Network Engineer | therefore you are Atlantic Net| _ http://www.lewis.org/~jlewis/pgp for PGP public key_ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Presence and IM?
Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PLEASE HELP X100P no responding
[EMAIL PROTECTED] ~]# modprobe zaptel[EMAIL PROTECTED] ~]# modprobe wcfxoZT_CHANCONFIG failed on channel 1: No such device or address (6)FATAL: Error running install command for wcfxo [EMAIL PROTECTED] ~]# ztcfg -vvv Zaptel Configuration==Channel map:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6) [EMAIL PROTECTED] /dev/zap]# ls -latotal 0drwxr-xr-x 2 root root 120 jun 17 15:45 .drwxr-xr-x 9 root root 5440 jun 17 15:45 ..crw-rw 1 asterisk asterisk 196, 254 jun 17 15:45 channelcrw-rw 1 asterisk asterisk 196, 0 jun 17 15:45 ctlcrw-rw 1 asterisk asterisk 196, 255 jun 17 15:45 pseudocrw-rw 1 asterisk asterisk 196, 253 jun 17 15:45 timer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp - fax problem
I only get the first 1cm of the page in tiff. Already tried to change the version of libtiff (.71), spandsp (pre18), asterisk (CVS) and nothing! The quality of image on that small band (1cm) is perfect. Miguel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Bjorn, Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. Bryan M. Johns One Ring Networks 300 West Wieuca Road, NE Building One Suite 205 Atlanta, GA 30342 404.303.9900 x: 104 http://www.oneringnetworks.com On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote: Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Bjorn, Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. Bryan M. Johns One Ring Networks 300 West Wieuca Road, NE Building One Suite 205 Atlanta, GA 30342 404.303.9900 x: 104 http://www.oneringnetworks.com On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote: Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone lookup
Quick question about call routing. We're currently setting up our system so that any phone calls made from our system over a t1 line to another legacy system go through a dedicated t1 server. Is there any method of checking to see if a number dialed exists on the system? Any help would be appreciated. Aaron Daniel Sr. Voice Analyst Sam Houston State University ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. No need to install AMP to get this, just install FOP : http://www.asternic.org/ hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple phones on a Zap FXS extension
Hi I have Asterisk up and running perfect with a Digium TDM400P card and 4 FXS ports. There are 4 ATT 4-Line 954 phones hooked to the system Each of the 4 lines is hooked to each phone. The problem is when you are on line 1 (or any line) and someone else picks up line 1 they can here the conversation. When the phone are hooked to the PSTN in the same way line 1 will be lite up to show that it is in use and you cannot join the call without pressing a certain key sequence. How can I get the phones to act like they do when they are connected to the PSTN when hooked to the Asterisk server. Any help is appreciated. Kelly ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729
On Fri, 2005-06-17 at 12:33 -0400, David wrote: Hi All, I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest Firmware) to use G729. In sip.conf I have set disallow=all, allow=g729 Please take the time to read the Sipura documentation where it states that you can only do ONE G729 call at a time on a SPA 1000, 2000 and 3000. The processor in the unit is not powerful enough to do 2 G729 calls. You have to allow ulaw or alaw so the other line can make a call while the first is busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: tdm400p not working after cvs-head update
I have tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks, today i did a CVS update to the latest head files and the card is not working. Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) HELP!. thanks -- David Romero## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 17, 2005, at 7:56 AM, Daryl G. Jurbala wrote: You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400 and I'll give you the amount I save over the next quarter. NPA-NXX is 215-862. Good luck. That sounds almost like Xeno's Paradox there... if you gave away the savings you still be paying the same amount thus half the savings would be...? Sorry, just had to inject some Friday afternoon humor onto the list. Seriously though, I was never able to get a T1 for that price anywhere myself until I moved to Orange County, CA. -Rob. -- Robert Goodyear | Managing Partner | Brand Up LLC 901 Calle Amanecer | Suite 150 | San Clemente, CA 92673 Tel: 949/468.0370 x501 | Fax: 949/468.0371 | Cell/SMS: 949/981.7301 http://brand-up.com | [EMAIL PROTECTED]___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP files for Polycom
Title: MGCP files for Polycom Does anybody know were I can download the MGCP files for the Polycom IP500? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: tdm400p not working after cvs-head update
It looks from here like you've rebooted the system after checkout, but your system was not configured to load zaptel drivers at boot time. Have you forgot to do 'make config' while in /usr/src/zaptel ? Hope this helps - Original Message - From: David Romero [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 17, 2005 11:30 PM Subject: [Asterisk-Users] Re: tdm400p not working after cvs-head update I have tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks, today i did a CVS update to the latest head files and the card is not working. Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) HELP!. thanks -- David Romero ## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callqueues confused :(
-- Started music on hold, class 'default', on SIP/193.111.200.67-0815c790 -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/101|20|tr) in new stack -- Called 101 -- Agent/1001 is ringing -- SIP/101-6fe1 is ringing -- Agent/1001 is ringing -- SIP/101-6fe1 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '' in context 'sip' -- Playing 'pbx-invalid' (language 'en') I have a call queue that appears to work until the agent is requested to press #. At this point it tries to transfer the call but then says the extension is invalid. Any pointers appreciated, I've drawn a blank. Thanks Neil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Saturday, 18 June 2005 2:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds cheapest in the world. Ha Ha Ha Ha How is your electricity sold? Hour, Watt, or by unit (KW/h)? And as for cell phones being cheap, you have a receiver pays setup! How good is that, then you have so many competing Telcos that sometimes you just cannot call the house across the street without tracversing 3 exchanges, and then you have so many area codes (each of which is so small) that almost all your calls are STD, then as for cheap, how is AUD$49.00/Month for all you can eat (that's about US$30/month) and all incoming calls are free, YES FREE! :) Really??? http://www.cucumber.com/fullinternational29.htm Look at the difference they (and everybody else) charges to call the cell network. Check your bill (your landline bill) next month and tell hoe much cell phone costs you. The point I was making is that the charges are NOT on _My_ cell phone bill, when I don't originate the call, however in .us if you get called, you pay, that can easily cost you a heap of money that you can only control by switching the phone off, and where is the point in that? So if I rec'v 500 calls a week on my cell phone, it still costs me nothing. And in some cases if I have the Cell and the Landline from the same telco (in .au), calls between them are free too, regardless of where I happen to be in australia at the time. Oh, and cucumber seem to be doing you no favours either I can place a call to the US using my Cell phone for 1-2c/minute, shrug Caviat Emptor? So much for the american(sic) always making things better. I knew that none americans might not see this unless pointed out to them :) Ha Ha Ha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400 and I'll give you the amount I save over the next quarter. NPA-NXX is 215-862. Good luck. Ive got a Full T1 from a rather large Mid-Atlantic CLEC for $291. Ive got about dozen of them from DC to Trenton, NJ. -Darren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ael files
Hi noticing the cvs updates of late, I'm wondering if there is support for fifo/shell commands in the extended dialplan language? can it fully replace agi scripts? Looks really interesting... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Rate Calculation
Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480 I may well be wrong but I believe the issue needs visiting. Somebody was asking me how it calculates costs as they thought they knew what a call should cost. I said I'll show you. Mistake, I could not come up with an answer that made sense. Please let me know, Darren Wiebe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm400p not working after cvs-head update
did you udate first? - Original Message - From: David Romero To: Asterisk-Users@lists.digium.com Sent: Friday, June 17, 2005 9:36 AM Subject: [Asterisk-Users] tdm400p not working after cvs-head update I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,today i did a CVS update to the latest head files and the card is not working.Zaptel Configuration== Channel map:Channel 01: FXS Kewlstart (Default) (Slaves: 01)Channel 02: FXS Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)4 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6)HELP!.thanks David Romero## ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callqueues confused :(
t in your dial statement? - Original Message - From: Neil Bullock [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 17, 2005 3:48 PM Subject: [Asterisk-Users] callqueues confused :( -- Started music on hold, class 'default', on SIP/193.111.200.67-0815c790 -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1001 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/101|20|tr) in new stack -- Called 101 -- Agent/1001 is ringing -- SIP/101-6fe1 is ringing -- Agent/1001 is ringing -- SIP/101-6fe1 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge -- Started music on hold, class 'default', on Local/[EMAIL PROTECTED],2 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '' in context 'sip' -- Playing 'pbx-invalid' (language 'en') I have a call queue that appears to work until the agent is requested to press #. At this point it tries to transfer the call but then says the extension is invalid. Any pointers appreciated, I've drawn a blank. Thanks Neil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue/How to get the number of incoming calls
Hi,all, Now,I am working at make an realtime monitorfor the call center based on asterisk. and ,I had search the archive and wiki.Through the return info from the management API, I canget the waiting calls ,abandoned calls ,hold time, etc,but I don't know how to get the number of incoming calls. The info like following : Asterisk Call Manager/1.0 Response: Success Message: Authentication accepted Response: Success Message: Queue status will follow Event: QueueParams Queue:test_queue Max: 18 Calls: 1 Holdtime: 127 Completed: 209 Abandoned: 18 ServiceLevel: 0 ServicelevelPerf: 0.0 Event: QueueMember Queue: test_queue Location: Agent/101 Membership: static Penalty: 0 CallsTaken: 20 LastCall: 1118794338 Event: QueueMember Queue: test_queue Location: Agent/102 Membership: static Penalty: 0 CallsTaken: 19 LastCall: 1118778909 Event: QueueMember Queue: test_queue Location: Agent/103 Membership: static Penalty: 0 CallsTaken: 14 LastCall: 1118782495 Event: QueueMember Queue: test_queue Location: Agent/104 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/105 Membership: sta tic Penalty: 0 CallsTaken: 9 LastCall: 1118779889 Event: QueueMember Queue: test_queue Location: Agent/106 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/107 Membership: static Penalty: 0 CallsTaken: 0 LastCall: 0 Event: QueueMember Queue: test_queue Location: Agent/108 Membership: static Penalty: 0 CallsTaken: 146 LastCall: 1118795077 Event: QueueEntry Queue: test_queue Position: 1 Channel: Zap/6-1 CallerID: 4042662907 Wait: 739 Response: Goodbye Message: Thanks for all the fish. *** who knows how to get it through such info ,or there are other methodfor getting incoming calls number? Any advice and help will be appreciated! Best Regards, Gary Li DO YOU YAHOO!? G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to find a path from g729 to gsm
Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's my account page but still i have the same error when I attempt to make a call. Second, my last digit is not allowed from teliax. that means I need one more digit from teliax for dialing through them. Third, I have somewhat poor support from teliax since I have send them 3,4 emails and so far i got no replies. Please, help me to go ahead from this point Thanking all of you, Kumara. The error I got Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm -- IAX2/teliax-1 is ringing -- IAX2/teliax-1 answered Zap/1-1 Jun 17 18:47:18 WARNING[7396]: channel.c:2308 ast_channel_make_compatible: No path to translate from Zap/1-1(68) to IAX2/teliax-1(256) Jun 17 18:47:18 WARNING[7396]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make Zap/1-1 compatible with IAX2/teliax-1 -- Hungup 'IAX2/teliax-1' == Spawn extension (outgoing, 19737228839, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Local numbers
If I set up an * server will I still be able to use my local Anchorage phone number through my * box? Thanks for any help, Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users