Re: [Asterisk-Users] Can an outside caller dial an extension before someone answer?
Short of making this time based or having multiple inbound numbers you cant do this without answering the call and reading dtmf (or as explained this last week T1/E1 lines may or may not be able to pass audio data incl dtmf for upto 90 seconds when it starts to ring). Now here is a problem, how would someone know to dial the extension if there is no automated attendant telling them to do so? Most callers wouldnt know to do this as most systems dont accept this. There is a trick for those 'in the know', asterisk answers the call but instead of playing a recorded voice it plays a ringing tone. Users would then be able to dial an extension if they know to do this. On Mon, 2005-09-26 at 16:48 +1200, Simon Glass wrote: Hi, We don't want a digital receptionist if we can help it (too impersonal!), but is it possible for an outside caller to dial an internal extension (eg 201) after asterisk answers the call, but before someone in the incoming call ring group has answered? -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change codec based on callerid (sip/iax)
I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser (www.iptel.org/ser) in between the asterisk box and forward effectivly to a different account on the asterisk box based on caller id (ie ser makes a choice which account to use). codecs then would be negotiated normally at connect time. -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problem setting up TDM22B card
Hi, I got the following output when I run the command. genzaptelconf -svd ./genzaptelconf: line 616: /etc/init.d/asterisk: No such file or directory Unloading zaptel modules: wcusb zaptel Test Loading modules: - zaptel - zaphfc - qozap Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wctdm Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxo Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcfxs - pciradio Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - tor2 Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - torisa Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct1xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wct4xxp Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg - wcte11xp - wcusb - ztd_eth Updating '/etc/default/zaptel' Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-channels.conf' Reconfiguring identified channels Zaptel Configuration == Channel map: 0 channels configured. ./genzaptelconf: line 653: /etc/init.d/asterisk: No such file or directory Checking channels configured in Asterisk: ./genzaptelconf: line 665: asterisk: command not found What may be the problem? Help me in this regard. Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 25, 2005 at 10:42:52PM -0700, somesh s wrote: Hi, Can you please give me some details about the link you have sent? I am not aware of what it does? [http://tzafrir.org.il/genzaptelconf] It is a bash script for generating zaptel.conf and zapata.conf according to the current settings. To use it: wget http://tzafrir.org.il/genzaptelconf bash genzaptelconf -h # gives help Try -s and -v . -d is probably not recommended if you have more thn one card, I figure. Regards, Somesh S. Shanbhag --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Sep 23, 2005 at 06:22:06AM -0700, somesh s wrote: Hi Steve, This is zaptel.conf. Can you please tell me if you require to see more conf files? [zaptel.conf] loadzone = us defaultzone=us fxoks=1-2 fxsks=3-4 http://tzafrir.org.il/genzaptelconf Should auto-detect zaptel.conf settings. Just in case you're not sure. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
this can help u: SIP.CONF [1] host = dynamic type = friend language = it qualify = no dtmfmode = rfc2833 callgroup = 1 pickupgroup = 1 callerid = "Bruno De Luca 1" 1 secret = 1234 mailbox = 1 context=1 [2] host = dynamic type = friend language = it qualify = no dtmfmode = rfc2833 callgroup = 2 pickupgroup = 2 callerid = "Bruno De Luca 2" 2 secret = 1234 mailbox = 2 context=2 [3] ... context=1 [4] ... context=2 EXTENSIONS.CONF [1] exten = 1,1,Dial(SIP/1) exten = 3,1,Dial(SIP/3) [2] exten = 2,1,Dial(SIP/2) exten = 4,1,Dial(SIP/4) trixter http://www.0xdecafbad.com wrote: They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP in Japan using Freebit
Has anyone had any experience using a VOIP provider in Japan? No matter what I try, my REGISTER string kicks back one of 2 errors: Got SIP response 481 Call/Transaction Does Not Exist back from x.x.x.x or Got SIP response 400 Bad Request back from x.x.x.x My register string is as follows: [EMAIL PROTECTED] I have tried the following also: 05075034132:[EMAIL PROTECTED] [EMAIL PROTECTED]/05075034132 05075034132:[EMAIL PROTECTED]/05075034132 myuserid:[EMAIL PROTECTED] and variations of the above. Is there any other information I could provide in order to get some help? I guess another thing I am looking for is a list of possible registration strings.. I'll try them all :D Cheers ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Check the BIOS options -- many others in the x3nn Series as well as the Netfinity before them allow you to specify the IRQ per slot through a deeply buried BIOS config option. I'm not near my rack of IBM servers to boot one to get the exact path but email me offline if you can't find it. g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MS Live Communication Server
Hi all, i have now managed to place a call from LCS to asterisk/pstn and it seems to work fine. Unfortunately i have still problems for incomming calls from asterisk/pstn to LCS. i have seen in the mailinglist that there seems to be problem calling from lcs to asterisk. Have anyone maneged to place a call from lcs to *. thx in advance... --- richard Coco [EMAIL PROTECTED] wrote: Hi, i have the same setup too. [exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20] Unfortunately i don't know how to configure the dialplan in my LCS. Can you please give me a hint where to configure this. thx. --- Jacky [EMAIL PROTECTED] wrote: LCS 2005 just support SIP TCP or TLS right now. so you must patch asterisk chan_sip.c support TCP, look http://bugs.digium.com/view.php?id=4903 I have successful call to asterisk's SIP peer or PSTN use Office Communicator 2005(sign-in my LCS 2005) but I can't use Dial(SIP/[EMAIL PROTECTED]) , let asterisk's SIP user invite LCS's user. Need any input. 2005/8/11, bubuk [EMAIL PROTECTED]: Hi List! does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heared about that, please let me know. The fact i figured out is that the Border Controler from Jasomi can be used as a gateway from MS-LCS-SIP to regular SIP. But that is not really handy and expensive too. Thank you Volker ___ 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
That doesnt really help. As stated in the email you replied to what is to prevent someone doing say [1] exten = 1,1,goto(2,1,1) or customer A *and* customer B trying to define the same context name, to use your example lets say they both want to create context '1'. I want to be able to create 1 system that has multiple users who are able to create their own dialplans without naming collisions with other customers or gotos going to other customers, etc. This is more for a virtual hosting type setup so I can have one large machine instead of many smaller ones, thus allowing for better ROI. While many have suggested that I learn the basics of contexts (as you did) no one has been able to ansewr the actual question asked making me think there is no current answer, and an AGI is the way to go. That way I can have more control over what data is observed and all that. I just didnt want to write an AGI if there was an existing solution, especially if it was part of asterisk itself and not an external program. On Mon, 2005-09-26 at 09:31 +0200, Bruno De Luca wrote: this can help u: EXTENSIONS.CONF [1] exten = 1,1,Dial(SIP/1) exten = 3,1,Dial(SIP/3) [2] exten = 2,1,Dial(SIP/2) exten = 4,1,Dial(SIP/4) trixter http://www.0xdecafbad.com wrote: They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy compile again
Kevin, I've got the source package in /usr/src/linux-2.6.5-1.358 I also have sym links to it from /usr/src/linux and /usr/src/linux-2.6 and a sym link /lib/modules/2.6.5-1.358/build to /usr/src/linux (as mentioned on voipinfo). If I ommit this last sym link then the complier complains about only needing the lib headers and not the full kernel (or something like that). Mark (I've checked and the .config is in the directory) Unless you can think of something obvious I'll clean out the src directory and redownload the kernel source and see if that helps. Kevin Collins wrote: Mark, Have you checked to make sure your kernel source is in the following directory : /usr/src/linux-2.6.5-1.358' Makefile:434: .config: No such file or directory It just seems to be complaining about not finding your kernel development source environment. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson Sent: Friday, September 23, 2005 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ztdummy compile again When you say kernel development do you mean kernel sources (which I have) or some other development tools/libs? and a kernel build config file? make mrproper ? make oldconfig ? I've done that much at least... Mark Kevin Collins wrote: Looks like you don't have kernel development installed and a basic kernel build config file generated. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson Sent: Friday, September 23, 2005 8:55 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztdummy compile again Hi, I'm still strugling with getting an easy to use conference system implemented. I did have app_conference running, but today I upgraded asterisk to 1.0.9 and it stopped working. I've tried following the instructions for compiling app_conference on 1.0.7 but it didn't work. So I went back to ztdummy (I've not had any luck getting this to compile on FC2). Anyhoo, I've tried again and once again ztdummy fails to compile and the various disparate instructions on what is needed to get it running are not helping. If I run make linux26 then the zaptel drivers start to compile but then spews out a load of errors. Anyone have any ideas? SNIP=== 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -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 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.5-1.358/build make -C /lib/modules/2.6.5-1.358/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.5-1.358' Makefile:434: .config: No such file or directory CC [M] /usr/src/zaptel/zaptel.o In file included from /usr/src/zaptel/zconfig.h:9, from /usr/src/zaptel/zaptel.c:40: include/linux/config.h:4:28: linux/autoconf.h: No such file or directory In file included from /usr/src/zaptel/zaptel.c:40: /usr/src/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory /usr/src/zaptel/zconfig.h:68:41: missing binary operator before token ( In file included from include/linux/kernel.h:11, from /usr/src/zaptel/zaptel.c:42: include/linux/linkage.h:5:25: asm/linkage.h: No such file or directory In file included from include/linux/types.h:13,
RE: [Asterisk-Users] IBM x306
Hi all, we have same problem with a x346. Mainly, TE410P shares IRQ with network card and if you change IRQ for this slot, automatically change IRQ in network card. Any idea? srsergio -Mensaje original- De: George Pajari [mailto:[EMAIL PROTECTED] Enviado el: lunes, 26 de septiembre de 2005 10:09 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] IBM x306 I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Check the BIOS options -- many others in the x3nn Series as well as the Netfinity before them allow you to specify the IRQ per slot through a deeply buried BIOS config option. I'm not near my rack of IBM servers to boot one to get the exact path but email me offline if you can't find it. g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 23/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
contact support at digium. I would doubt that your popping has anything to do with the rev of the card though; that's most likely an issue with shared interrupts (cat /proc/interrupts), and/or other motherboard resources. I am wondering -- I have a tdm400p with two modules and I understand that there is a REV I of the card --- Ihave e/f. I am getting somepopping and I definitely need echo cancel when bridged. Now would a rev I help in these matters and how can I get them to replace mine without or with minimal charge -- if anyone knows how this is supposed to work, please let me know. on Sunday 09/25/2005 Kevin P. Fleming([EMAIL PROTECTED]) wrote Rod Bacon wrote: Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking, etc.). Echo on bridged calls has also gone (I have now been able to disable echo cancellation on bridged calls, too!). Bridged calls with 2nd gen firmware result in the audio never leaving the card; that's why you are seeing such an improvement. Essentially, the Zaptel 'native bridge' is pushed all the way down into the card, so the audio stream is never passed across the PCI bus (it's not even packetized, just directly connected between the two channels). Glad to hear it was worth the time it took to get to you! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Registry problems
Hi there I am trying to register 2 servers using iax server a and server b server a registers to server b but when i say iax2 show registry i can see it is not using port 4569 xxx.xxx.xxx.xxx:4569 123456 xxx.xxx.xxx.xxx:1024 60 Registered and now i can't register server b to server aall ports are open on the router but still timeouts what i can do is use port 1024 to register to server a but that port is changing from time to time :( i am confused ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip, call ransfer and call waiting
Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Voip provider
www.viatalk.com is Asterisk friendly, and with purchase of a business plan (any of them) you can get a toll free number. With the addition of a toll free number, we allow multiple channels to be assigned to your account. This solves the problem of multiple inbound/outbound calls From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott WolfeSent: Sunday, September 25, 2005 9:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Best Voip provider All of the providers given so far seem to have a limited simultaneous connections. As a business solution (multiple outgoing calls at one time) what are you guys using? -Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip, call ransfer and call waiting
On Mon, 2005-09-26 at 11:08 +0200, Daniel ANDRE wrote: Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Yup... exten = 123,1,SetGroup(user1) exten = 123,2,CheckGroup(1) ; dont let more than 1 call at a time exten = 123,3,Dial(sip/user1) exten = 123,103,Busy ; this is where it goes if CheckGroup indicates more than X calls ... see http://voip-info.org/wiki-Asterisk+cmd+SetGroup for more info. You may have to play games with variables to make a macro perhaps that would be more generic in this regard, but this should at least get you started. -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage-type service
Yes - -Original Message- - From: [EMAIL PROTECTED] - [mailto:[EMAIL PROTECTED] On Behalf Of Waldo - Rubinstein - Sent: Sunday, September 25, 2005 9:06 PM - To: Asterisk Users Mailing List - Non-Commercial Discussion - Subject: [Asterisk-Users] Vonage-type service - - Is anyone offering a vonage-like service using a 100% asterisk only - solution? Just for curiosity. - - Thanks, - Waldo - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Date based context inclusion
Hi, I know that writing in the dialplan include = day|09:00-19:59|mon-fri|*|* day will be include monday TO friday What is needed to include day monday AND friday ? include = day|09:00-19:59|mon,fri|*|* does not work, but it was just my guess Tnx for any help -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Date based context inclusion
This should work include = day|09:00-19:59|mon|*|* include = day|09:00-19:59|fri|*|* -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Alessio Focardi Verzonden: maandag 26 september 2005 11:36 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Date based context inclusion Hi, I know that writing in the dialplan include = day|09:00-19:59|mon-fri|*|* day will be include monday TO friday What is needed to include day monday AND friday ? include = day|09:00-19:59|mon,fri|*|* does not work, but it was just my guess Tnx for any help -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Satellite Broadband and VOIP
Hi Sean, We operate a VSAT network here in the Philippines (using Shiron, FDMA Bandwidth on Demand) and offer VoIP using asterisk. We do not sell our voip to our gilat clients since gilat has a higher latency (since it uses TDMA). Try to look for a satellite provider that has an average (to your country of voip destination) latency of below 600-800 ms and it must be consistent. Also since, satellite has low upload bandwidth, try to have QoS behind the satellite modem and prioritize VoIP traffic cxpcman wrote: Sean Rima wrote: I live in a very rural area, BB access will never happen and the only choice I have it Satellite. I seen from a post to this list that Gilat sat modems are not recommended. Is this still the case or is there another alternative? Sean ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well is not recommended because of the seektime . the information you send and recive have a delay no matter how fast your conection is .. so you gonna hear the voice out of time . wire have a lot faster response times than air soo... ur choice ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Satellite Broadband and VOIP
What provider to use depends of course of witch country you live in. We have a lot of customers in Africa who use iwayafrica. Many of the providers block Voip because they have own Voip service. For US we use New Era Systems, Inc Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Anthony C. Delfin Sent: den 26 september 2005 12:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Satellite Broadband and VOIP Hi Sean, We operate a VSAT network here in the Philippines (using Shiron, FDMA Bandwidth on Demand) and offer VoIP using asterisk. We do not sell our voip to our gilat clients since gilat has a higher latency (since it uses TDMA). Try to look for a satellite provider that has an average (to your country of voip destination) latency of below 600-800 ms and it must be consistent. Also since, satellite has low upload bandwidth, try to have QoS behind the satellite modem and prioritize VoIP traffic cxpcman wrote: Sean Rima wrote: I live in a very rural area, BB access will never happen and the only choice I have it Satellite. I seen from a post to this list that Gilat sat modems are not recommended. Is this still the case or is there another alternative? Sean ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well is not recommended because of the seektime . the information you send and recive have a delay no matter how fast your conection is .. so you gonna hear the voice out of time . wire have a lot faster response times than air soo... ur choice ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing selected text with asterisk under windows ...
Hi list, I am looking for a windows application which extends the windows context menu to dial every selected string. I want to click on phone numbers on websites and let asterisk 1st ring my phone, when I pickup 2nd establish the call. Does anybody know such an app? Thank you Gerd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
On Monday 26 September 2005 00:36, Kevin P. Fleming wrote: Bridged calls with 2nd gen firmware result in the audio never leaving the card; that's why you are seeing such an improvement. Essentially, the Zaptel 'native bridge' is pushed all the way down into the card, so the audio stream is never passed across the PCI bus (it's not even packetized, just directly connected between the two channels). This is why so many of us are pushing Digium to PLEASE FOR THE LOVE OF GOD print a detailled list of what's improved with the new firmware... None of us have any clear idea of what has changed from v1 to v2 and little things like this are unbelievably important. Kind of like how some of us are also pushing for a more detailed changelog... not cvs log type of depth but Bugs fixed in this release: #105 #3033 #5050 etc and features added/removed/changed with a little more detail. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
On Monday 26 September 2005 00:43, Jean-Yves Avenard wrote: This new firmware only works on new hardware I guess.. The firmware is *on* the card itself. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls
I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the error I see below with receiving incoming calls? My asterisk box is behind my IPCop firewall. The current configuration works fine for outgoing calls, but has problems with receiving incoming ones. My current configuration looks like: [general] context=default bindaddr=192.168.0.4 srvlookup=no disallow=all allow=ulaw localnet=192.168.0.0/255.255.255.0 externip=65.87.XXX.XXX nat=no fromdomain = mydomain.com [200.XXX.XXX.XXX] type=peer secret=asterisk host=200.XXX.XXX.XXX allow=ulaw context=outgoing dtmfmode=rfc2833 insecure=very [from-200.XXX.XXX.XXX] type=user host=200.XXX.XXX.XXX allow=ulaw canreinvite=no context=outgoing insecure=very Outgoing calls seem to work fine, but there is no indication of any incoming calls in the SIP debug information when I call the DID number externally. I have all the SIP and RTP port forwarded to my Asterisk box in my firewall and don't see anything in the firewall logs. I do see the following 2 entries back-to-back in an ethereal dump. I don't know enough about SIP to know if the DID side is sending a bad INVITE or if Asterisk is not handling the INVITE correctly. I cannot tell if the DID side is not responding back with more address detail or if my Asterisk box is dropping the connection right after the 484 response. Can someone help? Thanks, Frank No. TimeSourceDestination Protocol Info 2497 21.504651 XXX-IPA.155.115.200.in-addr.arpa lyla.mydomain.com SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description Frame 2497 (1088 bytes on wire, 1088 bytes captured) Arrival Time: Sep 22, 2005 23:19:50.962763000 Time delta from previous packet: 0.003659000 seconds Time since reference or first frame: 21.504651000 seconds Frame Number: 2497 Packet Length: 1088 bytes Capture Length: 1088 bytes Ethernet II, Src: 00:04:e2:bc:76:80, Dst: 00:0e:0c:62:cb:08 Destination: 00:0e:0c:62:cb:08 (lyla.mydomain.com) Source: 00:04:e2:bc:76:80 (SmcNetwo_bc:76:80) Type: IP (0x0800) Internet Protocol, Src Addr: XXX-IPA.155.115.200.in-addr.arpa (200.115.155.XXX), Dst Addr: lyla.mydomain.com (192.168.0.4) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 00.. = Differentiated Services Codepoint: Default (0x00) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 1074 Identification: 0x (0) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 44 Protocol: UDP (0x11) Header checksum: 0x2632 (correct) Source: XXX-IPA.155.115.200.in-addr.arpa (200.115.155.XXX) Destination: lyla.mydomain.com (192.168.0.4) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: XXX4 Checksum: 0x3933 (correct) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 200.115.155.XXX:5060 Via: SIP/2.0/UDP 200.115.155.XXX:5061;branch=z9hG4bK-e4907aa1 From: office1 sip:[EMAIL PROTECTED];tag=bc58fe6c90fb9969o1 SIP Display info: office1 SIP from address: sip:[EMAIL PROTECTED] SIP tag: bc58fe6c90fb9969o1 To: sip:[EMAIL PROTECTED] SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 69 Contact: office1 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-agent: Sipura/SPA3000-2.0.10(GWf) Content-Length: 432 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Record-Route: sip:200.115.155.XXX:5060;lr Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 13054566 13054566 IN IP4 200.115.155.XXX Owner Username: - Session ID: 13054566 Session Version: 13054566 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 200.115.155.XXX Session Name (s): - Connection Information (c): IN IP4 200.115.155.XXX Connection Network Type: IN Connection
[Asterisk-Users] Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls
I'm new to asterisk and need some help with getting a SIP connection
working.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
register my SIP connection in order to send or receive calls.
Can someone help me with how to understand the error I see below with
receiving incoming calls?
My asterisk box is behind my IPCop firewall. The current configuration
works fine for
outgoing calls, but has problems with receiving incoming ones.
My current configuration looks like:
[general]
context=default
bindaddr=192.168.0.4
srvlookup=no
disallow=all
allow=ulaw
localnet=192.168.0.0/255.255.255.0
externip=65.87.XXX.XXX
nat=no
fromdomain = mydomain.com
[200.XXX.XXX.XXX]
type=peer
secret=asterisk
host=200.XXX.XXX.XXX
allow=ulaw
context=outgoing
dtmfmode=rfc2833
insecure=very
[from-200.XXX.XXX.XXX]
type=user
host=200.XXX.XXX.XXX
allow=ulaw
canreinvite=no
context=outgoing
insecure=very
Outgoing calls seem to work fine, but there is no indication of any
incoming calls in the SIP debug
information when I call the DID number externally. I have all the SIP and
RTP port forwarded to my Asterisk box in my firewall and don't see
anything in the firewall logs.
I do see the following 2 entries back-to-back in an ethereal dump. I
don't know enough about
SIP to know if the DID side is sending a bad INVITE or if Asterisk is not
handling the INVITE
correctly. I cannot tell if the DID side is not responding back with more
address detail or if my Asterisk box is dropping the connection right
after the 484 response.
Can someone help?
Thanks,
Frank
No. TimeSourceDestination Protocol
Info
2497 21.504651 XXX-IPA.155.115.200.in-addr.arpa lyla.mydomain.com
SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session
description
Frame 2497 (1088 bytes on wire, 1088 bytes captured)
Arrival Time: Sep 22, 2005 23:19:50.962763000
Time delta from previous packet: 0.003659000 seconds
Time since reference or first frame: 21.504651000 seconds
Frame Number: 2497
Packet Length: 1088 bytes
Capture Length: 1088 bytes
Ethernet II, Src: 00:04:e2:bc:76:80, Dst: 00:0e:0c:62:cb:08
Destination: 00:0e:0c:62:cb:08 (lyla.mydomain.com)
Source: 00:04:e2:bc:76:80 (SmcNetwo_bc:76:80)
Type: IP (0x0800)
Internet Protocol, Src Addr: XXX-IPA.155.115.200.in-addr.arpa
(200.115.155.XXX), Dst Addr: lyla.mydomain.com (192.168.0.4)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
00.. = Differentiated Services Codepoint: Default (0x00)
..0. = ECN-Capable Transport (ECT): 0
...0 = ECN-CE: 0
Total Length: 1074
Identification: 0x (0)
Flags: 0x04 (Don't Fragment)
0... = Reserved bit: Not set
.1.. = Don't fragment: Set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 44
Protocol: UDP (0x11)
Header checksum: 0x2632 (correct)
Source: XXX-IPA.155.115.200.in-addr.arpa (200.115.155.XXX)
Destination: lyla.mydomain.com (192.168.0.4)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Source port: 5060 (5060)
Destination port: 5060 (5060)
Length: XXX4
Checksum: 0x3933 (correct)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 200.115.155.XXX:5060
Via: SIP/2.0/UDP 200.115.155.XXX:5061;branch=z9hG4bK-e4907aa1
From: office1 sip:[EMAIL PROTECTED];tag=bc58fe6c90fb9969o1
SIP Display info: office1
SIP from address: sip:[EMAIL PROTECTED]
SIP tag: bc58fe6c90fb9969o1
To: sip:[EMAIL PROTECTED]
SIP to address: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 69
Contact: office1 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 432
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Record-Route: sip:200.115.155.XXX:5060;lr
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 13054566 13054566 IN IP4
200.115.155.XXX
Owner Username: -
Session ID: 13054566
Session Version: 13054566
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 200.115.155.XXX
Session Name (s): -
Connection Information (c): IN IP4 200.115.155.XXX
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 200.115.155.XXX
RE: [Asterisk-Users] Will Digium Wildard work with PCI-Xor PCI Express
I had trouble with a TE110P card in a Supermicro mobo - P8SCT. The PRI line kept dropping calls when the card was in a standard PCI slot. In the end the only way to fix it was to install the card in the PCI-X slot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth Sent: 22 September 2005 21:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Will Digium Wildard work with PCI-Xor PCI Express Just correcting myself. The 3 PCI-X slots are one 64-bit 133 MHz and two 64-bit 100 MHz. Matt Matt Roth wrote: Don't bank on it. We were going to use a Wildcard as a timing source on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on the PCI-X slot I installed it in, sometimes the box wouldn't even boot. For perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots (one 64-bit 133 MHz, two 32-bit 100 MHz). I believe the timing is only needed for music on hold, IAX trunking, and MeetMe conferencing. We're not doing trunking or conferencing (for now) so we're going with ztdummy. If the timing isn't perfect only our music on hold will suffer, which is no big deal. If we run into other problems, we might try popping our quad-span card in there just to see if it works. Keep in mind that Digium no longer produces Wildcards. I'm not sure why they don't work with our 6850 and the techs at Dell didn't know either. Maybe they are not 100% PCI compliant. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Kevin Bockman wrote: Chuck Bunn wrote: Does anyone know if the Digium Wildcard will work on a PCI Express or PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack server for use with Asterisk. They will work in PCI-X of course but not PCI Express. They are totally different. You will need the 3.3v cards. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
On Monday 26 September 2005 00:43, Jean-Yves Avenard wrote: This new firmware only works on new hardware I guess.. The firmware is *on* the card itself. -A. How can I check to see if I have the new firmware or not? I bought a card used the other day. Thanks, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
On 9/26/05, Steve Totaro [EMAIL PROTECTED] wrote: How can I check to see if I have the new firmware or not?I bought a cardused the other day. Type dmesg | grep TE410P version and look for the TE410P version line. If it ends with 164 you got the new firmware. [EMAIL PROTECTED] ~]# dmesg | grep TE410P versionTE410P version c01a010b, burst ON-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Registry problems
Good reading for you: http://lists.digium.com/pipermail/asterisk-users/2005-March/097986.html I trust you are going out over a NAT and the machines are not on the same subnet. Thanks, Steve - Original Message - From: Sander To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, September 26, 2005 2:03 AM Subject: [Asterisk-Users] IAX Registry problems Hi there I am trying to register 2 servers using iax server a and server b server a registers to server b but when i say iax2 show registry i can see it is not using port 4569 xxx.xxx.xxx.xxx:4569 123456 xxx.xxx.xxx.xxx:1024 60 Registered and now i can't register server b to server aall ports are open on the router but still timeouts what i can do is use port 1024 to register to server a but that port is changing from time to time :( i am confused ___--Bandwidth and Colocation sponsored by Easynews.com --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 No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pager Notification Script
Tom Rymes a écrit : Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to Asterisk). Then, if that message hasn't been retrieved in 5 minutes, I want to send another page. The same goes after 10 and 15 minutes. After 20 minutes, I want to send another page *AND* send an e-mail or generate a call to another party. Off Site Notification or Off Premise Notification... I have write a script that is part of ScopServ but here how it work: - Create per-user configs using GUI (ex. after 10 min send to a voicemail, after 20 min. send to a pager, etc) (email, pager, voicemail) - Use externnotify in voicemail.conf - If # of msg = 0 then delete all pending notification else - Retreive per-user config and check action - Create action in a second table with timestamp + x min. - A crontab that check at each minute for action, execute if and delete the row in table. - Create .call file or send email -- Joel Vandal ScopServ Inc. http://www.scopserv Inc. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help)
Hi, Yes, I have changed it. I tried as you have done, I put txgain and rxgain to 0and it was possible receive and send faxes. I'm usingtwo TDM400P (FXS with 4 ports and FXO with 4 ports). In my tests I'm sending faxes to asterisk from fax machine connected to one of the FXS ports. The problem is, if put tx and rx gain to 0in the conversations comingfromFXO channels hear very verylow. What can I do then, any idea ? On 9/23/05, Chris [EMAIL PROTECTED] wrote: Are you trying through Zap channels?Have you changed the RXGain or TXGain? I can send faxes if I use RXGain= 20.0 but I can not receive unless Ihave the RX and TX set to 0.Chris- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Friday, September 23, 2005 2:15 PMSubject: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi,I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax machine side. my extensions.conf exten =301,2,Background(mp) exten =fax,1,RxFax(/home/admin/testfax.tif) and I have tried with as well. Press * star on the fax machine and after hear the fax tone press the start button to send the document. exten =301,2,Background(mp) exten =*,1,RxFax(/home/admin/testfax.tif) Can somebody help me ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best drivers for HFC-S ISDN cards
It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help)
Turn up the volume on your phones? - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 26, 2005 6:12 AM Subject: Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi, Yes, I have changed it. I tried as you have done, I put txgain and rxgain to "0"and it was possible receive and send faxes. I'm usingtwo TDM400P (FXS with 4 ports and FXO with 4 ports). In my tests I'm sending faxes to asterisk from fax machine connected to one of the FXS ports. The problem is, if put tx and rx gain to 0in the conversations comingfromFXO channels hear very verylow. What can I do then, any idea ? On 9/23/05, Chris [EMAIL PROTECTED] wrote: Are you trying through Zap channels?Have you changed the RXGain or TXGain? I can send faxes if I use RXGain= 20.0 but I can not receive unless Ihave the RX and TX set to 0.Chris- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Friday, September 23, 2005 2:15 PMSubject: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi,I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax machine side. my extensions.conf exten =301,2,Background(mp) exten =fax,1,RxFax(/home/admin/testfax.tif) and I have tried with as well. Press * star on the fax machine and after hear the fax tone press the start button to send the document. exten =301,2,Background(mp) exten =*,1,RxFax(/home/admin/testfax.tif) Can somebody help me ? ___--Bandwidth and Colocation sponsored by Easynews.com --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 No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: [Asterisk-Users] Vonage-type service
I want to share some facts with the Asterisk community. I have been very successful providing a Vonage-type system based on Asterisk. For instance, one company that uses Asterisk and offers a similar service to Vonage is Voyze.com. The key concept is that Asterisk works like a Cisco, for all the intelligence is provided by SQL Server, outside Linux. I don't even save the CDR locally. The configuration files, like sip.conf, are downloaded from SQL Server, where they are generated and modified by triggers that execute in several tables. The Management GUI is simply an application that modifies SQL tables, and so does the Web application, for the end customer. Both are written with Microsoft Visual Studio 2003. It works perfectly, is scalable and very cheap to maintain. I use freetds and UnixODBC to link both worlds, Linux and Windows 2003 Enterprise. We don't sell the system. We provide a full independent system for customers including co-location, for a setup fee and 1/2 cent per call, regardless of length. We also provide US termination via our own DS3 for 1.3 cents a minute, and it does support T.38 faxing. Federico Alves ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: [Asterisk-Users] Vonage-type service
I dont know about others, but i find that SER as a SIP proxy in front of asterisk works much better for endpoints on the public internet than just asterisk as a sip proxy. my 2 cents. -yair On 9/26/05, Federico Alves [EMAIL PROTECTED] wrote: I want to share some facts with the Asterisk community. I have been verysuccessful providing a Vonage-type system based on Asterisk. For instance, one company that uses Asterisk and offers a similar service to Vonage isVoyze.com. The key concept is that Asterisk works like a Cisco, for all theintelligence is provided by SQL Server, outside Linux. I don't even save the CDR locally. The configuration files, like sip.conf, are downloaded from SQLServer, where they are generated and modified by triggers that execute inseveral tables. The Management GUI is simply an application that modifies SQL tables, and so does the Web application, for the end customer. Both arewritten with Microsoft Visual Studio 2003. It works perfectly, is scalableand very cheap to maintain. I use freetds and UnixODBC to link both worlds, Linux and Windows 2003 Enterprise.We don't sell the system. We provide a full independent system for customersincluding co-location, for a setup fee and 1/2 cent per call, regardless oflength. We also provide US termination via our own DS3 for 1.3 cents aminute, and it does support T.38 faxing.Federico Alves___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Back On Busy?
I know it's been
touched on before, but no answers have been found to the best of my knowledge.
I'm using a SIP only setup, with a sip provider giving PSTN and would like to
see if anyone has an idea for creating redial busy using ${DIALSTATUS} and
possibly MeetMe?
I figure something
like this, but want to get feedback
1. Get callers last
dialed number, if international number, do not allow.
2. Playback a
stuttertone to caller
3. Disconnect
caller
4. Ring intended
party check dial status. If busy, wait120 seconds and try again (do this
for a total of 15 minutes)
5. If it's picked
up, playback an announcement to the party and put them in a meetme
conference
6. Ring the original
caller and bridge them to the meetme conference.
___
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[Asterisk-Users] Re: goiax expanded with free us domestic calling
Joe Stewart wrote: On Fri, Sep 23, 2005 at 11:12:24AM -0700, Matthew Simpson wrote: I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Thank you very much for setting up this service. I've successfully made calls, but unlike my other iax trunks the callerid isn't passed on so the call comes in from areacode 202. Any hints to get this working? The caller ID thing is intended behavior. Passing the 87820-xxx number doesn't usually show up so it will come up as the 202 number. Currently the site hands out a virtual 87820-xxx number but I intend to add the ability to get a free United States DID [possibly other countries as well] as well. Please test it out. You can use an IAXy, asterisk, or an IAX softphone like iaxcomm. I've only used asterisk. If I have a chance I'll try a softphone. Any chance of g729? I know that since this is iax your options would be more limited as far as licensing. GSM is available. It takes up far less CPU than G729 and is about equal in quality and bandwidth usage. just wanted to send you a note and say thanks, Joe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Dying Signal 11
Try analyzing the core file, it should be in the /tmp dir. If you need to further debug do a backtrace on the core file. At the Linux command type gdb asterisk corefilename , this should give you some information and pinpoint the culprit. http://www.voip-info.org/tiki-index.php?page=Asterisk+debugging Alberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Friday, September 23, 2005 3:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk - Dying Signal 11 Hi, Asterisk keeps dying reporting error signal 11. There is no segmentation fault etc and full logging reports nothing with respect to reasons of why it restarts. Any ideas? Regards, Sahil Gupta VoiceValley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Back On Busy?
This may not apply to your situation, but
many ATAs and SIP phones have this feature built in to the device.
We use Linksys/Sipura and auto redial and
last call return work without any special setup.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Monday, September 26, 2005
7:45 AM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call
Back On Busy?
I know it's been touched on before, but no answers have been
found to the best of my knowledge. I'm using a SIP only setup, with a sip
provider giving PSTN and would like to see if anyone has an idea for creating
redial busy using ${DIALSTATUS} and possibly MeetMe?
I figure something like this, but want to get feedback
1. Get callers last dialed number, if international number,
do not allow.
2. Playback a stuttertone to caller
3. Disconnect caller
4. Ring intended party check dial status. If busy,
wait120 seconds and try again (do this for a total of 15 minutes)
5. If it's picked up, playback an announcement to the party
and put them in a meetme conference
6. Ring the original caller and bridge them to the meetme
conference.
___
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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] Best drivers for HFC-S ISDN cards
It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? CAPI doesn't work with this card. mISDN should work, but zaphfc should prolly be fine as well. isdn4linux will probably work, and give you a long term headache, nausea and possibly hemroids. you may want to tune the echo cancellation in zaptel (echocancel=256 or something) to see if that helps... roy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help)
I have already to the highest volume. I have tried with different phones models and the same results... On 9/26/05, Steve Totaro [EMAIL PROTECTED] wrote: Turn up the volume on your phones? - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 26, 2005 6:12 AM Subject: Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi, Yes, I have changed it. I tried as you have done, I put txgain and rxgain to 0and it was possible receive and send faxes. I'm usingtwo TDM400P (FXS with 4 ports and FXO with 4 ports). In my tests I'm sending faxes to asterisk from fax machine connected to one of the FXS ports. The problem is, if put tx and rx gain to 0in the conversations comingfromFXO channels hear very verylow. What can I do then, any idea ? On 9/23/05, Chris [EMAIL PROTECTED] wrote: Are you trying through Zap channels?Have you changed the RXGain or TXGain? I can send faxes if I use RXGain= 20.0 but I can not receive unless Ihave the RX and TX set to 0.Chris- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Friday, September 23, 2005 2:15 PM Subject: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi,I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax machine side. my extensions.conf exten =301,2,Background(mp) exten =fax,1,RxFax(/home/admin/testfax.tif) and I have tried with as well. Press * star on the fax machine and after hear the fax tone press the start button to send the document. exten =301,2,Background(mp) exten =*,1,RxFax(/home/admin/testfax.tif) Can somebody help me ? ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 -- Angel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Back On Busy?
Thank you, I do appreciate that many ATAs have redial on
busy, but I've been given the charge of figuring out how one would do it in
Asterisk.
Don't ask me why
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: Monday, September 26, 2005 10:15 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Call Back On Busy?
This may not apply to
your situation, but many ATAs and SIP phones have this feature built in to the
device.
We use Linksys/Sipura
and auto redial and last call return work without any special
setup.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: [Asterisk-Users] Call Back On
Busy?
I know it's been touched on
before, but no answers have been found to the best of my knowledge. I'm using
a SIP only setup, with a sip provider giving PSTN and would like to see if
anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly
MeetMe?
I figure something like this, but
want to get feedback
1. Get callers last dialed number,
if international number, do not allow.
2. Playback a stuttertone to
caller
3. Disconnect
caller
4. Ring intended party check dial
status. If busy, wait120 seconds and try again (do this for a total of
15 minutes)
5. If it's picked up, playback an
announcement to the party and put them in a meetme
conference
6. Ring the original caller and
bridge them to the meetme conference.
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] Re: goiax expanded with free us domestic calling
How long do you plan on your service remaining free? Joe Stewart wrote: On Fri, Sep 23, 2005 at 11:12:24AM -0700, Matthew Simpson wrote: I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Thank you very much for setting up this service. I've successfully made calls, but unlike my other iax trunks the callerid isn't passed on so the call comes in from areacode 202. Any hints to get this working? The caller ID thing is intended behavior. Passing the 87820-xxx number doesn't usually show up so it will come up as the 202 number. Currently the site hands out a virtual 87820-xxx number but I intend to add the ability to get a free United States DID [possibly other countries as well] as well. Please test it out. You can use an IAXy, asterisk, or an IAX softphone like iaxcomm. I've only used asterisk. If I have a chance I'll try a softphone. Any chance of g729? I know that since this is iax your options would be more limited as far as licensing. GSM is available. It takes up far less CPU than G729 and is about equal in quality and bandwidth usage. just wanted to send you a note and say thanks, Joe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CheckGroup accross multiple servers
I'm running multiple asterisk servers and need to use the CheckGroup function (and other group functions) across multiple servers Ex: - there are 5 channels in group test on server 1 - there are 8 channels in group test on server 2 I would need a checkgroup to return me 13. Any way to currently do this ? What would be the best way to implement this if not ? Store group setting in shared mysql? Thanks Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best drivers for HFC-S ISDN cards
On Mon, 26 Sep 2005, Roy Sigurd Karlsbakk wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? CAPI doesn't work with this card. mISDN should work, but zaphfc should prolly If mISDN does work, then CAPI (chan_capi) can be used for this card too, because mISDN provides a CAPI interface. Armin be fine as well. isdn4linux will probably work, and give you a long term headache, nausea and possibly hemroids. you may want to tune the echo cancellation in zaptel (echocancel=256 or something) to see if that helps... roy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help)
I dont know if it will work or not but have you tried setting each card up with different gains? I am not sure if the variables are global or not, but when I installed an FXO card on an [EMAIL PROTECTED] box and used genzaptelconf, it created a separate file for the FXO with its own rx settings and was included in zaptel.conf. - Original Message - From: [EMAIL PROTECTED] To: Steve Totaro ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, September 26, 2005 7:28 AM Subject: Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help) I have already to the highest volume. I have tried with different phones models and the same results... On 9/26/05, Steve Totaro [EMAIL PROTECTED] wrote: Turn up the volume on your phones? - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 26, 2005 6:12 AM Subject: Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi, Yes, I have changed it. I tried as you have done, I put txgain and rxgain to "0"and it was possible receive and send faxes. I'm usingtwo TDM400P (FXS with 4 ports and FXO with 4 ports). In my tests I'm sending faxes to asterisk from fax machine connected to one of the FXS ports. The problem is, if put tx and rx gain to 0in the conversations comingfromFXO channels hear very verylow. What can I do then, any idea ? On 9/23/05, Chris [EMAIL PROTECTED] wrote: Are you trying through Zap channels?Have you changed the RXGain or TXGain? I can send faxes if I use RXGain= 20.0 but I can not receive unless Ihave the RX and TX set to 0.Chris- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Friday, September 23, 2005 2:15 PM Subject: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi,I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax machine side. my extensions.conf exten =301,2,Background(mp) exten =fax,1,RxFax(/home/admin/testfax.tif) and I have tried with as well. Press * star on the fax machine and after hear the fax tone press the start button to send the document. exten =301,2,Background(mp) exten =*,1,RxFax(/home/admin/testfax.tif) Can somebody help me ? ___--Bandwidth and Colocation sponsored by Easynews.com --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 No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 -- Angel ___--Bandwidth and Colocation sponsored by Easynews.com --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 No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best drivers for HFC-S ISDN cards
On Mon, Sep 26, 2005 at 02:23:04PM +0100, Chris Bagnall wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. mISDN (kernel modules and user lib) is used by chan_misdn and chan_capi. vISDN might be another option. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). There are some Audio Quality Tuning Options for zap channels. What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? chan_misdn, chan_capi (echosquelch) and vISDN include a very basic echo cancellation or non echo cancellation. IMHO bristuff is the best (least worse) choice. chan_modem/isdn4linux will cause additional delay and will itensify the echo problem. Do not use reply if you want to start a new thread. The header In-Reply-To: is used by thread-aware mail clients even if the subject has been changed. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help)
I haven't found a solution to it.It's not a matter of turning up the phone volume because I'm receiving and sending the faxes directly from Asterisk with SpanDSP (Pre 20).I've tried different versions of SpanDSP and LibTiff. It's the same result no matter what. Regards, Chris - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 26, 2005 6:12 AM Subject: Re: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi, Yes, I have changed it. I tried as you have done, I put txgain and rxgain to 0 and it was possible receive and send faxes. I'm using two TDM400P (FXS with 4 ports and FXO with 4 ports). In my tests I'm sending faxes to asterisk from fax machine connected to one of the FXS ports. The problem is, if put tx and rx gain to 0 in the conversations coming from FXO channels hear very very low. What can I do then, any idea ? On 9/23/05, Chris [EMAIL PROTECTED] wrote: Are you trying through Zap channels? Have you changed the RXGain or TXGain? I can send faxes if I use RXGain= 20.0 but I can not receive unless I have the RX and TX set to 0. Chris - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, September 23, 2005 2:15 PM Subject: [Asterisk-Users] Can't receive Faxes with Asterisk (help) Hi, I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax machine side. my extensions.conf exten =301,2,Background(mp) exten =fax,1,RxFax(/home/admin/testfax.tif) and I have tried with as well. Press * star on the fax machine and after hear the fax tone press the start button to send the document. exten =301,2,Background(mp) exten =*,1,RxFax(/home/admin/testfax.tif) Can somebody help me ? -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/05 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax problem
On Sun, Sep 25, 2005 at 07:26:12AM -0600, Rich Adamson wrote: Two approaches that have been rather common are: 1. use the separate contexts for each did, 2. in the register statement, add /1234 at the end; like register = username:[EMAIL PROTECTED]/6789 I don't think it will work , iax statement don't have exten on end. [..] register user[:password] @ remote_host [:port] To register with another IAX server. [..] This is true for SIP but not for IAX. For #2, incoming calls would be handled with: exten = 6789,1,Dial(SIP/1235) Besides that : *CLI iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered X.X.X.X:4569 Username2 [MYIP]:456960 Registered X.X.X.X:4569 Username3 [MYIP]:456960 Registered source and destination ports for all 3 iax registrations are the same , and my isp see only one, becouse rest is overwriten. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Hi: I am running AAH and setup Broadvoice, but when I call in to the BV number I cannot send dial commands to my auto attendant or speak if I use a did to send the inbound calls to a specific extension. I'll gladly capture an SID debug and place a call, or post any necessary conf files. TIA Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I cannot send dial commands to my auto attendant or speak if I use a did to send the inbound calls to a specific extension Does asterisk says something in the verbose console? please post your sip.conf relevant entries for BroadVoice. I have just cancelled with BroadVoice (too much latency for the places i wanted to call), so i never used the incoming number. But im glad to help if i can. Best RegardsOn 9/26/05, Jason Schafer [EMAIL PROTECTED] wrote: Hi:I am running AAH and setup Broadvoice, but when I call in to the BVnumber I cannot send dial commands to my auto attendant or speak if Iuse a did to send the inbound calls to a specific extension.I'll gladly capture an SID debug and place a call, or post any necessary conffiles.TIAJason___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recent Sphinx integration work?
Hi all, I know this has been covered previously in the lists, but I was wondering if anyone has some recent experience integrating Asterisk (preferable 1.2 beta 1) and Sphinx (version 3.5 or 4)? I haven't found too much information on how to actually do so. If you have done this, or have some pointers on where to look (I haven't turned up too many helpful examples via Google), please feel free to contact me at jhosteny at mac dot com. Thanks, Joe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI ISDN on USB
Hi, I would like to add a BRI ISDN line to my AMP box but I cant use a PCI card because there are no PCI slots. I have seen that some USB ISDN modems works with CAPI drivers and maybe with Asterisk. Does someone have already connected a BRI ISDN line to Asterisk using a USB adapter and what models can I use (French ISDN compatible)? Regards, Amaury ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Best drivers for HFC-S ISDN cards
Can anyone get zaphfc to work with CVS HEAD? The version currently available from Junghans (f) is against CVS HEAD as of 2005-05-29 - it doesn't patch very well on the latest CVS HEAD. -- Vidar - Original Message - From: Stefan Tichy [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, September 26, 2005 4:54 PM Subject: [Asterisk-Users] Re: Best drivers for HFC-S ISDN cards On Mon, Sep 26, 2005 at 02:23:04PM +0100, Chris Bagnall wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. mISDN (kernel modules and user lib) is used by chan_misdn and chan_capi. vISDN might be another option. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). There are some Audio Quality Tuning Options for zap channels. What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? chan_misdn, chan_capi (echosquelch) and vISDN include a very basic echo cancellation or non echo cancellation. IMHO bristuff is the best (least worse) choice. chan_modem/isdn4linux will cause additional delay and will itensify the echo problem. Do not use reply if you want to start a new thread. The header In-Reply-To: is used by thread-aware mail clients even if the subject has been changed. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax problem
Two approaches that have been rather common are: 1. use the separate contexts for each did, 2. in the register statement, add /1234 at the end; like register = username:[EMAIL PROTECTED]/6789 I don't think it will work , iax statement don't have exten on end. [..] register user[:password] @ remote_host [:port] To register with another IAX server. [..] Ops... This is true for SIP but not for IAX. For #2, incoming calls would be handled with: exten = 6789,1,Dial(SIP/1235) Besides that : *CLI iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered X.X.X.X:4569 Username2 [MYIP]:456960 Registered X.X.X.X:4569 Username3 [MYIP]:456960 Registered source and destination ports for all 3 iax registrations are the same , and my isp see only one, becouse rest is overwriten. Have you tried using three different contexts for those in iax.conf? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk vertical service activation codes
Here's what I learned (thanks Larry): For example, assume you're paying for three way calling from your telco, and you want to place a three way call. First establish a call with person A. Press the Flash (or Link) button. When you get the dialtone from Asterisk, press *0 and Asterisk sends the flashhook to the CO. Now you hear another dialtone, this time from the CO. Now dial the number for person B. Press Flash (or link again), then press *0 to flash the CO again. Now you have established a three way call through the telco. This only works with Zap channels of course. Hugh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Does asterisk says something in the verbose console? I'm not sure what the verbose console is, but I can run sip debug and post the output when I make an inbound call. please post your sip.conf relevant entries for BroadVoice. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pedantic=no ; added for Broadvoice support 8/3/05 EK externip=216.xxx.xxx.xxx localnet=172.xxx.xxx.0/255.255.255.0 I have just cancelled with BroadVoice (too much latency for the places i wanted to call), so i never used the incoming number. But im glad to help if i can. I have outbound setup on VOIPJet, my intent with the Broadvoice is to setup a forward on busy with my landline to roll over to the BV number. Here's the output from sip debug m=audio 14008 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 13 headers, 12 lines Using latest request as basis request Sending to 147.135.0.128 : 5060 (non-NAT) Found no matching peer or user for '147.135.0.128:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 101 Peer audio RTP is at port 147.135.0.128:14008 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for s in from-sip-external list_route: hop: sip:[EMAIL PROTECTED]:5060;ep=147.135.0.129;transport=udp Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack -- Executing Goto(SIP/147.135.0.129-095da350, from-pstn|s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(SIP/147.135.0.129-095da350, ) in new stack We're at 216.xxx.xxx.xxx port x Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 238 v=0 o=root 1782 1782 IN IP4 216.xxx.xxx.xxx s=session c=IN IP4 216.xxx.xxx.xxx t=0 0 m=audio 14138 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 ACK Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Max-Forwards: 69 Content-Length: 0 9 headers, 0 lines -- Executing SetVar(SIP/147.135.0.129-095da350, intype=aa_2) in new stack -- Executing Cut(SIP/147.135.0.129-095da350, intype=intype|-|1) in new stack -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?13:15) in new stack -- Goto
Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express
Chuck, This is my mistake. I thought that only the X100P was dubbed a Wildcard. All of my posts regarding this subject are specific to the X100P. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Chuck Bunn wrote: Hi, You stated that Digium is discontinuing the Wildcard series - that would be there whole product line! In particular I am looking at the Wildcard TDM 400P series of cards.. Thanks Matt Roth wrote: Don't bank on it. We were going to use a Wildcard as a timing source on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on the PCI-X slot I installed it in, sometimes the box wouldn't even boot. For perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots (one 64-bit 133 MHz, two 32-bit 100 MHz). I believe the timing is only needed for music on hold, IAX trunking, and MeetMe conferencing. We're not doing trunking or conferencing (for now) so we're going with ztdummy. If the timing isn't perfect only our music on hold will suffer, which is no big deal. If we run into other problems, we might try popping our quad-span card in there just to see if it works. Keep in mind that Digium no longer produces Wildcards. I'm not sure why they don't work with our 6850 and the techs at Dell didn't know either. Maybe they are not 100% PCI compliant. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Kevin Bockman wrote: Chuck Bunn wrote: Does anyone know if the Digium Wildcard will work on a PCI Express or PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack server for use with Asterisk. They will work in PCI-X of course but not PCI Express. They are totally different. You will need the 3.3v cards. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 way calling?
Here's what I learned (thanks Larry): For example, assume you're paying for three way calling from your telco, and you want to place a three way call. First establish a call with person A. Press the Flash (or Link) button. When you get the dialtone from Asterisk, press *0 and Asterisk sends the flashhook to the CO. Now you hear another dialtone, this time from the CO. Now dial the number for person B. Press Flash (or link again), then press *0 to flash the CO again. Now you have established a three way call through the telco. This only works with Zap channels of course. Repeat the above on the second line to get the 5 way activated... Hugh On 8/17/05, hugolivude [EMAIL PROTECTED] wrote: I'd not bother with using the flash based 3 way calling. Instead I'd setup an account with an ITSP and make the outbound calls via IP, preferabbly via IAX2. That way to can reach out to as many people as your bandwidth allows. Simply. Conveniently. Add one IP based DID and you can let others call in to your conference via IP. I've been thinking about getting some IP DIDs for other reasons anyway, so thanks for the suggestion. There's a bandwidth issue however and this client is simply more comfortable keeping things on copper, especially con-calls. As I mentioned the client's paying for 3-way calling from Bell, so is there no way to take advantage of this and establish a three way call on a single FXO line through Asterisk? I've opened another thread on this issue as it's more fundamental than my original 5 way calling problem. Thanks, Hugh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: channel offhook state
Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject: channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 File: ATT00068.txt -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/2005 attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to CCM
I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: channel offhook state
Hi, Yes, I had the same. Incoming calls were fine it was just when I made outgoing calls the line would sometimes hang and I would get all circuits are busy. Putting a butt (test) phone on the line in parallel indicated the line had dropped back to an on hook state, although asterisk wouldn't use it for some time. 20 mins. In the log it showed an error indicating it could not create a ZAP channel when I tried to create an outbound line. In the end I had to remove the card from the PC, run * without the card and run genzaptelconf to remove the zap-auto entries. I also removed all the outbound routing and removed by 4 ZAP trunks from the configs. I then shutdown the machine and re-installed the card and let * find the hardware and then re-ran genzaptelconf again. Im sure there is another more appropriate solution, but im an * newbie and I was clutching at straws!!! Regards Matt _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacqueline Lee Sent: 26 September 2005 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: channel offhook state Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject: channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 File: ATT00068.txt -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/2005 File: ATT00080.txt attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN
Hello, I want to send oH323 calls toour Quintum D3000. I have installed oH323 but I need a working sample oh323.conf and extensions.conf, so that I can route specific calls to the Quintum using H323. For example our Asterisk box IP=192.168.10.100 and Quintum IP=192.168.10.101. Can anyone assist with a sample Extensions.conf and oH323.conf. Thank you, Ade. Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with voicemail ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: channel offhook state
FWIW, there were a couple of channel zap changes made in the last couple of days to cvs-head. Don't have a clue whether those fixes addressed the problem you're talking about. Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject:[Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject:channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 File: ATT00068.txt -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/2005 ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Need Help on Areski Calling Card Solution plz
Hi Uppal, Thanks for you response. I was able to work on the gui and all the database things but I where I think I got lost is creating trunk, tarrifgroup, ratecard. I will be glad if u can throw more lights on this section for me. Thank you for your reply Goksie On 9/25/05, Junaid Uppal [EMAIL PROTECTED] wrote: AreskiCC works great for me , i've been using it for ~ 500 + cards scene and it works awesome for me! really , the guy did a REALLY good job , trust me. cheers ~uppal On 9/25/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: My experience with Areski is I wasn't able to get it to work and wasn't able to get help including from the owner of idiot guide who inturns wasn't able to get areski to work either according to him. I easily downloaded astcc and works fine Regards; Chawki Hammoud --- ADEGOKE ARUNA [EMAIL PROTECTED] wrote: Can someone share its working files experience on areskicc with me. I got it installed but my sip user and iax could not get registered talkless of making call and all the include directives instructed in the idiot guide were followed. Can someone share its experience with me on this? Aruna -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CM Rahman Jr. Sent: Tuesday, July 19, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Comments on Areski Calling Card Solution plz I am using it. I liked it. The guy did a good job. He doesn't have the agent module yet. But I think that is on its way. Thanks Quoting Arnd Vehling [EMAIL PROTECTED]: Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ___ 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 CM Rahman Jr. CTO CCS Internet www.ccsi.com http://www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.comhttp://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 way calling?
Does this require any special changes or additions to features.conf, or does Asterisk by default know to flash the ZAP channel when *0 is pushed? Tom On Sep 26, 2005, at 12:11 PM, hugolivude wrote: Here's what I learned (thanks Larry): For example, assume you're paying for three way calling from your telco, and you want to place a three way call. First establish a call with person A. Press the Flash (or Link) button. When you get the dialtone from Asterisk, press *0 and Asterisk sends the flashhook to the CO. Now you hear another dialtone, this time from the CO. Now dial the number for person B. Press Flash (or link again), then press *0 to flash the CO again. Now you have established a three way call through the telco. This only works with Zap channels of course. Repeat the above on the second line to get the 5 way activated... Hugh On 8/17/05, hugolivude [EMAIL PROTECTED] wrote: I'd not bother with using the flash based 3 way calling. Instead I'd setup an account with an ITSP and make the outbound calls via IP, preferabbly via IAX2. That way to can reach out to as many people as your bandwidth allows. Simply. Conveniently. Add one IP based DID and you can let others call in to your conference via IP. I've been thinking about getting some IP DIDs for other reasons anyway, so thanks for the suggestion. There's a bandwidth issue however and this client is simply more comfortable keeping things on copper, especially con-calls. As I mentioned the client's paying for 3-way calling from Bell, so is there no way to take advantage of this and establish a three way call on a single FXO line through Asterisk? I've opened another thread on this issue as it's more fundamental than my original 5 way calling problem. Thanks, Hugh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pager Notification Script
Does this mean that you do have such a script but can't/won't share it? Tom On Sep 26, 2005, at 8:57 AM, Joel Vandal wrote: Tom Rymes a écrit : Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to Asterisk). Then, if that message hasn't been retrieved in 5 minutes, I want to send another page. The same goes after 10 and 15 minutes. After 20 minutes, I want to send another page *AND* send an e-mail or generate a call to another party. Off Site Notification or Off Premise Notification... I have write a script that is part of ScopServ but here how it work: - Create per-user configs using GUI (ex. after 10 min send to a voicemail, after 20 min. send to a pager, etc) (email, pager, voicemail) - Use externnotify in voicemail.conf - If # of msg = 0 then delete all pending notification else - Retreive per-user config and check action - Create action in a second table with timestamp + x min. - A crontab that check at each minute for action, execute if and delete the row in table. - Create .call file or send email -- Joel Vandal ScopServ Inc. http://www.scopserv Inc. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: channel offhook state
On Monday 26 September 2005 13:32, Rich Adamson wrote: FWIW, there were a couple of channel zap changes made in the last couple of days to cvs-head. Don't have a clue whether those fixes addressed the problem you're talking about. Don't think so, but they hardlock the kernel with te4xxp cards. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma and Digium same machine?
Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to CCM
Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk% 20Cisco%20CallManager%20Integration ? I've followed these steps and I can make calls from a CCM client to Asterisk, but the end point at the Asterisk side can't hear any audio. On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote: I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arnaldo M. Pereira egghunt at gmail dot com http://ansi-c.org/~arnaldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tux/Asterisk logo for Cisco phones
El jue, 22-09-2005 a las 19:04, David McNett escribió: I made http://slacker.com/~nugget/stuff/asterisk-cow-real.bmp for my non-Linux asterisk servers. I made my * + tux + office logo http://www.cipher.com.pe/central/asterisk-tux-cipher.bmp Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I am also a long time client, and have no incoming BV today. -Darren From: [EMAIL PROTECTED] on behalf of Jason Schafer Sent: Mon 9/26/2005 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice Does asterisk says something in the verbose console? I'm not sure what the verbose console is, but I can run sip debug and post the output when I make an inbound call. please post your sip.conf relevant entries for BroadVoice. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pedantic=no ; added for Broadvoice support 8/3/05 EK externip=216.xxx.xxx.xxx localnet=172.xxx.xxx.0/255.255.255.0 I have just cancelled with BroadVoice (too much latency for the places i wanted to call), so i never used the incoming number. But im glad to help if i can. I have outbound setup on VOIPJet, my intent with the Broadvoice is to setup a forward on busy with my landline to roll over to the BV number. Here's the output from sip debug m=audio 14008 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 13 headers, 12 lines Using latest request as basis request Sending to 147.135.0.128 : 5060 (non-NAT) Found no matching peer or user for '147.135.0.128:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 101 Peer audio RTP is at port 147.135.0.128:14008 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for s in from-sip-external list_route: hop: sip:[EMAIL PROTECTED]:5060;ep=147.135.0.129;transport=udp Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack -- Executing Goto(SIP/147.135.0.129-095da350, from-pstn|s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/147.135.0.129-095da350, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(SIP/147.135.0.129-095da350, ) in new stack We're at 216.xxx.xxx.xxx port x Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 238 v=0 o=root 1782 1782 IN IP4 216.xxx.xxx.xxx s=session c=IN IP4 216.xxx.xxx.xxx t=0 0 m=audio 14138 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 147.135.0.128:5060 -- Executing Wait(SIP/147.135.0.129-095da350, 1) in new stack asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr From: Schafer Trish sip:[EMAIL PROTECTED];user=phone;tag=SD28clb01-1612693231-1127750324179 To: Jason Schafersip:[EMAIL PROTECTED];user=phone;tag=as2a994d31 Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002 CSeq: 160704490 ACK Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Max-Forwards: 69 Content-Length: 0 9 headers, 0 lines -- Executing SetVar(SIP/147.135.0.129-095da350, intype=aa_2) in new stack -- Executing Cut(SIP/147.135.0.129-095da350, intype=intype|-|1) in new stack
RE: [Asterisk-Users] FW: channel offhook state
Yes indeed. There have been huge changes to chan_zap.c in CVS-HEAD compared to 1.09. In 1.09 Stable there are a lot of problems with handling call hang-ups. CVS-HEAD, of 28/08 was much better. But even though it did improve things, it wasn't quite right. In particular I found two problems with polarity reversal detection in chan_zap.c for which I have created a patch (this is now in CVS-HEAD). Please see http://bugs.digium.com/view.php?id=5191 for more details. Please note that you'll need to use answeronpolarityswitch=yes and/or hanguponpolarityswitch=yes in your Zapata.conf to make full use of the polarity detection code. You will also need to be very careful if CID is sent on a polarity switch too -- you may need to make it detect on the 0th ring or you could suffer from immediate hang-ups on ring. Unfortunately I've received a problem report with this modification. Any updates Magnus? I'm hoping it is all down to the ring that CID is detected at, and that by changing it to 0 or 1 all will be well again. But anybody who has had problems with hangup detection in the past should try CVS-HEAD and play with the options above to see if it improves things. Having said all this, things are still not perfect: For UK (and possibly other European countries) we still require a way for Asterisk to detect the continuous tone that indicates a remote party hangup on a POTS line. The Sipura 3000 uses this method and I believe it works quite well, though I've not tried it myself. Faris. -Original Message- FWIW, there were a couple of channel zap changes made in the last couple of days to cvs-head. Don't have a clue whether those fixes addressed the problem you're talking about. Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject:[Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject:channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] change codec based on callerid (sip/iax)
This can be done by modifying the source code. trixter http://www.0xdecafbad.com wrote: I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser (www.iptel.org/ser) in between the asterisk box and forward effectivly to a different account on the asterisk box based on caller id (ie ser makes a choice which account to use). codecs then would be negotiated normally at connect time. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN
Hi Ade,
An example of oh323.conf is attached, and the lines in the extensions.conf
that make the choice os this oh.323 channel is:
[globals]
GK = OH323/IP of your GK
[local]
ignorepat = 0
exten = _0021NXXX,1,Dial(${GK}/${EXTEN:1})
exten = _0021NXXX,2,Congestion
--
[ ]'s
Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
www.tecnologiaip.com.br
On Monday 26 September 2005 13:33, Ade Agbero wrote:
Hello,
I want to send oH323 calls to our Quintum D3000.
I have installed oH323 but I need a working sample oh323.conf and
extensions.conf, so that I can route specific calls to the Quintum using
H323.
For example our Asterisk box IP=192.168.10.100 and Quintum
IP=192.168.10.101.
Can anyone assist with a sample Extensions.conf and oH323.conf.
Thank you,
Ade.
-
Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with
voicemail
;
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; rtp.conf
;
udpStart=1
udpEnd=2
;
; Enable fast start (yes,no).
;
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=3
libTraceLevel=3
;libTraceFile=stdout
libTraceFile=/var/log/asterisk-h323.log
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is the zone
name.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; gatekeeper's DNS name,
; gatekeeper's ip,
; GKID:gatekeeper's id
; gatekeeper's id@gatekeeper's name or address
;
gatekeeper=IP of your gatekeeper
;gatekeeper=DISABLE
;
; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
; Q931- Q.931 Keypad Information Element
; STRING - H.245 string
; TONE- H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=voip-h323
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in all-aliases context.
;
context=all-aliases
alias=asterisk
alias=99001701
alias=99001702
;
;
; Aliases/prefixes routed in more-aliases context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in all-prefixes context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in more-stuff context.
;
;context=more-stuff
;alias=664
;gwprefix=02
;[cisco2]
;type=h323
;e164=02124950937
;context=all-aliases
;-
[Asterisk-Users] goiax caller ID
I'm not sure what he/she was sending as the caller ID information, what I was trying to do, was send a normal 10 digit number as caller ID. Is there any solution to this? Or anything planned? Thanks for your time, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ? In CLI not working
Has anyone noticed that a ? Entered at the root CLI does not work any longer? Petty I know but I did use it. --john -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
I have been trying on and off for a couple of weeks to no avail... Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IptablesAsterisk
I have Asterisk server(1.0.9) behind Iptables firewall. I configured Iptables and sip.conf as below. Andrea(2000) is the outsider phone, on Internet with public IP Luca(2001) is the insider phone, on local network with private IP as well Asterisk server. I noted the ports in play are 5060, 8000, 8001 and 1:2,so to test I put the large rule $IPTABLES -A FORWARD -p udp --dport 8000:2 -j ACCEPT Andrea or Luca receive the rings,but not the voice. Can you help me thank Andrea --- IPTABLES #!/bin/sh IPTABLES=/sbin/iptables # Internal network # LOC_IFACE=eth0 LOC_ADDR=10.100.0.0/24 LOC_IF=10.100.0.1 # External network # EST_IFACE=eth1 EST_ADDR=250.xxx.yyy.24/255.255.255.252 EST_IF=250.xxx.yyy.26 # Asterisk IP and port # PORAST=5060 ASTERISK=10.100.0.225 # deny everything for now # $IPTABLES -P INPUT DROP $IPTABLES -P FORWARD DROP $IPTABLES -P OUTPUT DROP # SIP on UDP port 5060 # $IPTABLES -A FORWARD -i $EST_IFACE -p udp -d $ASTERISK --dport $PORAST -m state --state NEW,ESTABLISHED -j ACCEPT $IPTABLES -A FORWARD -o $EST_IFACE -p udp -s $ASTERISK --sport $PORAST -m state --state ESTABLISHED -j ACCEPT # Other port for phone comunication # $IPTABLES -A FORWARD -p udp --dport 8000:2 -j ACCEPT # Allow from internal to external # $IPTABLES -A FORWARD -o $EST_IFACE -s $LOC_ADDR -m state --state NEW,ESTABLISHED -j ACCEPT $IPTABLES -A FORWARD -i $EST_IFACE -d $LOC_ADDR -m state --state ESTABLISHED -j ACCEPT $IPTABLES -t nat -A POSTROUTING -o $EST_IFACE -j SNAT --to $EST_IF # Asterisk on Internet # $IPTABLES -t nat -A PREROUTING -p udp -d $EST_IF --dport $PORAST -j DNAT --to $ASTERISK:$PORAST --- SIP.CONF [general] port = 5060 bindaddr = 0.0.0.0 allow = all context = bogon-calls [2000] type = friend username = 2000 callerid = Andrea Bencini 2000 secret = 9overthruster7 host = dynamic nat = yes context = from-sip mailbox = 100 [2001] type = friend username = 2001 callerid = Luca Bencini 2001 secret = 11bbanzai9 host = dynamic nat = yes context = from-sip mailbox = 101 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early Media in 180 Ringing
Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? U 10.254.254.1:5060 - 192.168.0.173:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35. Record-Route: sip:[EMAIL PROTECTED]:5060. Record-Route: sip:[EMAIL PROTECTED]:5060;lr;nat=yes. From: 0161801019 sip:[EMAIL PROTECTED];tag=as02de1b95. To: sip:[EMAIL PROTECTED];tag=00-04094-52dbe3bc-6cf68a723. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: sip:212.241.48.70:5060. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. # U 192.168.0.173:5060 - 192.168.1.103:5062 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265. From: 411 sip:[EMAIL PROTECTED];tag=f93ee2f65c6906cb. To: sip:[EMAIL PROTECTED];tag=as675f246d. Call-ID: [EMAIL PROTECTED] CSeq: 60590 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: sip:[EMAIL PROTECTED]. Content-Length: 0. . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] goiax caller ID
Kevin Scott wrote: I'm not sure what he/she was sending as the caller ID information, what I was trying to do, was send a normal 10 digit number as caller ID. Is there any solution to this? Or anything planned? There are no plans to allow just any caller ID to be sent. Once US dids are available than the DID cid would be sent instead. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording
Waldo, Thanks for the information. If you don't mind answering: are you guys developing this solution for your internal needs (meaning serving UAs from within your enterprise) or are you planning on offering services to the public? This solution is being developed for our internal needs. It's not that I'm really interested in your business or business model. I'm mainly curious to know how you will deal with potential UAs that are behind external NATs. Will you Asterisk farm stand behind a NAT or will it all be publicly accessible where no NAT translation or port forwarding will exist? I read the section on Asterisk and NAT on the wiki but still left me with some open questions. Our SIP traffic will never leave our internal network. There will be no NAT/firewalls to traverse. Calls to/from the PSTN will pass through a Cisco AS5400HPX Universal Gateway that handles the TDM/VoIP translation. In my particular setup, I work in a small call center. I have Asterisk behind one NAT with port forwarding on port 5060 and ports 1-2, only because I have 2 remote agents. The rest of the agents are in-house. The remote agents themselves are behind other NATs (behind their DSL service provider). Some times my Asterisk queues have trouble contacting the remote agents. At first, I thought I could simply put a SER server on the public edge, but I'm not sure if that will really solve the problem. I question this setup in terms of stability and security. Even worse, what would happen if my boss decides to increase the remote agents? I spoke to you privately about this and suggested using the IAX protocol with IAXy devices, but you indicated you needed to use SIP. Since we are not dealing with remote agents in our implementation, that is really all I can offer. I hope that the list members will be able to help you solve your problem. Sincerely, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to CCM
Are you using CCM to operate your gateway with MGCP? If so, I had to change the default timers under CCM advanced setup for Media exchange timers or the call was timing out at 4 seconds. If the setup was complete prior, it worked fine, but after 4 seconds q.931 from CCM would tear down the call.. On Mon, 2005-09-26 at 14:14 -0300, Arnaldo M. Pereira wrote: Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk% 20Cisco%20CallManager%20Integration ? I've followed these steps and I can make calls from a CCM client to Asterisk, but the end point at the Asterisk side can't hear any audio. On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote: I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma and Digium same machine?
Are you using Digium's new v2 firmware? If not I would recommend against it. I currently have 2 Sangoma quad T1 cards in a single server and it works just fine. Previously I had 2 TE405P(with old firmware) in the machine and had interrupt issues. Replaced with Sangoma boards before Digium v2 firmware was released to fix the problem. Haven't tried 2 Digium quad cards in single system yet. MATT--- On 9/26/05, William Lloyd [EMAIL PROTECTED] wrote: Anybody ever put a Sangoma and a Digium card in the same server?Specifically a four port card from each company?-bill[EMAIL PROTECTED]___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension availabilty
I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance, -Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Message Waiting Indicator (MWI) for remote voice mail?
I haven't received any responses. Just wanted to follow up and see if anyone has ideas? It seems like there ought to be a way to do this, especially since the TDM400 FXS card is able to send the proper signal to the connected phone. It seems like there just needs to be a way to configure the FXO card to pass though or bridge that signal/information to the FXS card when it is received at the FXO card. VOIP VM - Sipura - Phone worked. Asterisk VM - FXS - Phone works. Just need a way to do: VOIP VM - Sipura - FXO - ??? - FXS - Phone The above works great for everything else I've tried so far except for passing through the message waiting indicator. Thanks for any ideas! On 9/24/05, Brian McEntire [EMAIL PROTECTED] wrote: I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone. It also gives a broken dial tone. This is achieved by specifying mailbox=mb# in zapata.conf and possibly also by specifing adsi=yes in the same file. The question I have is this: I also have voicemail with my VOIP provider. Before jumping into Asterisk, the VOIP provider could send the message waiting indicator to my phone when I had new messages. After putting Asterisk between my analog handset and the VOIP adapter, the message waiting indicator from the VOIP provider seems to no longer get through to the phone. The connection to the VOIP provider is Cable Modem - Sipura 3002 - TDM FXO interface - TDM FXS interface - phone. Is there a way for Asterisk to get notified and pass the message waiting indicator on to my handset when there is a voice mail waiting at the VOIP provider? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Back On Busy?
Anyone else out there have some thoughts? The customer
wants to be able to control what can be redialed on busy, such as no
international. I'm having my doubts as to whether or not this can be done. My
idea seems like it would work, but after the customer hangs up, wouldn't the
context stop processing?
Thanks,
SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: Monday, September 26, 2005 10:15 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Call Back On Busy?
This may not apply to
your situation, but many ATAs and SIP phones have this feature built in to the
device.
We use Linksys/Sipura
and auto redial and last call return work without any special
setup.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: [Asterisk-Users] Call Back On
Busy?
I know it's been touched on
before, but no answers have been found to the best of my knowledge. I'm using
a SIP only setup, with a sip provider giving PSTN and would like to see if
anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly
MeetMe?
I figure something like this, but
want to get feedback
1. Get callers last dialed number,
if international number, do not allow.
2. Playback a stuttertone to
caller
3. Disconnect
caller
4. Ring intended party check dial
status. If busy, wait120 seconds and try again (do this for a total of
15 minutes)
5. If it's picked up, playback an
announcement to the party and put them in a meetme
conference
6. Ring the original caller and
bridge them to the meetme conference.
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RE: [Asterisk-Users] WRT54GP2 SIP server on LAN port
Thanks for the information Sherwood.
Then the question I had if the normal routing works for the SIP proxy
works with a LAN server.
But I cant get a success in connecting the router LINE1 to Asterisk.
WRT54GP2 says as status Can't connect to login server and there is no
connection attempt when running sip debug with verbose 4.
In my sip.conf this is specified:
[linksys]
type=friend
host=dynamic
username=100
secret=x
canreinvites=no
context=outgoing-sip
And in extensions.conf
[default]
exten = s,1,Dial(SIP/linksys|30|gr)
exten = s,2,VoiceMail(u100)
exten = s,3,Congestion
[outgoing-sip]
exten = _[0-9#*].,1,Dial(SIP/blixtvik-sip/${EXTEN}||t)
Now incoming calls gets the following loggs:
-- Executing Dial(SIP/0755xx-5499, SIP/linksys|30|gr) in new
stack
Sep 26 19:55:34 NOTICE[5525]: app_dial.c:777 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail(SIP/0755xxx-5499, u100) in new stack
-- Playing 'vm-theperson' (language 'se')
-- Playing 'digits/1' (language 'se')
== Spawn extension (default, s, 2) exited non-zero on
'SIP/0755xxx-5499'
Sep 26 19:55:37 ERROR[5525]: cdr_sqlite.c:136 sqlite_log: cdr_sqlite:
attempt to write a readonly database
Sep 26 19:55:37 ERROR[5525]: cdr_csv.c:222 csv_log: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
The answering machine works but it will not get connected with my WRT54GP2.
See anything that causes WRT54GP2 not to be able to register to Asterisk?
~Johannes
Actually, just point the line you want to use to a local ip address (the
asterisk server). I currently do this with my service. i.e. If your
Asterisk
server is 192.168.15.200, just make the proxy for line 1 that address. It
routes internally just fine.
Sherwood McGowan
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Sunday, September 25, 2005 5:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] WRT54GP2 SIP server on LAN port
what I do is loopback the WAN port to a LAN port and am able to use both
(ie) take a cable from the wan port of the router and plug it into the lan
port on the same router. This will give you a local ip and it still
should
allow connection out to your other provider.
On 9/25/05, Johannes [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I'm trying to set up Asterisk behind my WRT54GP2 router that has a
intergrated ATA box.
My box are not locked in any way so I can access and change all settings.
Now to the problem...
I have gotten Asterisk to register with my provider and everything works
just well..
Now it's time to get the intergrated ATA to connect to asterisk.
But the asterisk box in located on the LAN ports of the WRT54GP2.
I can't get the router to connect to Asterisk.
The question is then if the router does not use the normal routing table
and will force the connect to the SIP gateway to the WAN port even that I
specified a LAN IP as the gateway.
Has anyone set up the WRT54GP2 to connect to a asterisk server thats on
the LAN ports with a LAN IP? Or is this impossible?
Regards,
~Johannes
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] asterisk SMS and sprintpcs
Does anyone know about sending SMS messages to a sprint pcs phone. Can you give me a few details. Thanks, Jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VOIP in Japan using Freebit
Have you tried: [EMAIL PROTECTED]:[EMAIL PROTECTED] [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED] [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/05075034132 ? Sometimes SIP providers require the realm in the username, so the first part should have the @blah Then, the third part, is the callerid so it shouldn;t be required, and the last part, is the extension notification or something like that, I never use it. Always include the pass. Regards! Alchaemist Pikoro [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Has anyone had any experience using a VOIP provider in Japan? No matter what I try, my REGISTER string kicks back one of 2 errors: Got SIP response 481 Call/Transaction Does Not Exist back from x.x.x.x or Got SIP response 400 Bad Request back from x.x.x.x My register string is as follows: [EMAIL PROTECTED] I have tried the following also: 05075034132:[EMAIL PROTECTED] [EMAIL PROTECTED]/05075034132 05075034132:[EMAIL PROTECTED]/05075034132 myuserid:[EMAIL PROTECTED] and variations of the above. Is there any other information I could provide in order to get some help? I guess another thing I am looking for is a list of possible registration strings.. I'll try them all :D Cheers ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Carrier Access - Access Bank I config
Hi, Is there somebody using an Access Bank I with Asterisk that could share the secret ingredients needed to make it work ? I've searched around and found some info, I tryed almost every configuration possible but I can't seem to find the right combination. If someone could provide me with the config needed on Asterisk as well as the dip-switch settings on the channel bank part, I would be really greatfull. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sipuras 841 bad sound
Re: sipuras 841 bad sound (Juan Jose Comellas) On Tuesday 20 September 2005 20:46, Anton Krall wrote: I have a problems with some sipuras 841 and asterisk 1.0.9. (upgrade the firmware was suggested and completed, and didn't fix the problem.) There are a few little configuration details which are hard to catch on the SPA-841, which can affect sound quality. * RTP packet size: 0.20 On the SIP tab of the Advanced Admin page, the RTP packet size is shown, measured in seconds. It defaults to 0.03, however Asterisk is hardcoded to use 0.02. This mismatch can cause sound issues. * Silence Supp Enable: Off On the Ext1 and Ext2 tabs of Advanced Admin, the Silence Supp Enable option must be turned off. This is Silence Suppression, which causes the phone to stop sending RTP packets when the phone detects silence in the handset. Asterisk 1.0.9 does not support silence suppression, so this option must be turned off, or audio stream timing will fail a lot. We have a bunch of SPA-841's in service, and we're just finishing working out the bugs in the system. Our latest audio issue, as far as we can tell, was caused by a Duplex Mismatch between the ethernet port on the Asterisk server, and the ethernet port on the switch it was connected to. When one is set to full duplex and the other half duplex, you get random, intermittant periods of massive packet loss/jitter, which messes up audio something fierce. I've found http://www.voiptroubleshooter.com/ to be a good source of info on diagnosing random audio is bad issues. It has sound clips of the different kinds of audio is bad problems, along with info on what might cause that kind of problem. Alan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ring requested on channel already in use
I posted this 1.2.0-beta1 success story to asterisk-dev, and someone recommended that asterisk-users might benefit from it as well. Thanks, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] -- Forwarded message -- Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT) Subject: [Asterisk-Dev] Re: Ring requested on channel already in use To: asterisk-dev@lists.digium.com alan wrote: A problem was recently posted on the Asterisk-Users mailing list, and it went unresolved. Now that it's plaguing our production system as well, I need to look into it further. Good report, lots of information. See if you can reproduce it in CVS-HEAD (Asterisk, libpri, zaptel) snip You need to test this with cvs head (1.2beta) first to see if it's not already fixed... I am happy to say that since we upgraded to 1.2.0-beta1, our problems with Asterisk instability have not recurred. Our uptime is over a week, with the last restart a result of the upgrade. Thanks! I didn't like to see the answer upgrade your production system to a beta version, but the truth is, it was working poorly enough that it was basically impossible not to at least try it. Here is a summary of the symptoms we were seeing in 1.0.9, for others with this issue who may benefit from an upgrade: We narrowed the problem down to this sequence of events: - an incoming Zap call on a PRI channel - was sent to the queue - and answered by a AgentCallbackLogin queue agent - who was using a SIP phone - and the agent attempted to SIP REFER transfer the call - to another AgentCallbackLogin agent on a SIP phone That's a lot of channels (zap - agent - local - sip, transferring to agent - local - sip). When this happened, we saw these symptoms: - Rarely, the transfer succeeded. - More often, the ZAP channel was put in limbo and both SIP parties were dropped; or the transfer completed but there was one-way audio from Zap to SIP only. - Often, when the transfer failed, Asterisk was left in an inconsistent state, and would not function correctly until a restart was performed. -- asterisk -r consoles could not execute commands successfully -- sip show channels produced bogus output -- incoming Zap calls (over a PRI) resulted in Ring requested on channel... already in use errors, and the calling party was dropped immediately. After this experience with 1.2, I'd say that the upgrade should not cause many problems, as long as you thoroughly research and implement all required configuration changes. We have not experienced any problems with 1.2 which weren't also problems in 1.0.8/9, but we have had many other little issues solved which we were previously trying to ignore. Thank you very much, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension availabilty
The snom360 phones along with the current CVS-HEAD of Asterisk can presently do this. You'll want to do a wiki search on Hint in the dialplan for implementation details. Polycom has also just released a DSS sidecar to go with their 601 model phones, but the firmware to support more than 8 appearances at a time is still in the works. If you need something now, I'd go with snom360's and Asterisk. I have deployed this already in production and it is working quite well. The DSS LED lights solid when the person is on the line, and blinks when their phone is ringing with an incoming call. On 9/26/05, Joshua Laroff [EMAIL PROTECTED] wrote: I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance,-Josh___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren it works here today but they can be a bit unpredictable I use a cheap byod lite account mostly as a test tool. I figure if they grow up someday I might use them more. I have been wondering if they will meet the FCC deadlines or just fade away. At least some providers have been sending notices and collecting street addresses last few months. Others look like they are not really preparing to stay in the business when the deadlines hit. Maybe they are hoping another provider will buy the customer base and DID's? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension availabilty
FOP does this quite nicely From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua LaroffSent: Monday, September 26, 2005 1:57 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Extension availabilty I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated.Thanks in advance,-Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension availabilty
On Snom phones this feature works (look at the Hint Command in extension.conf.) Support for this should come for the Grandstream GXP2000, currently it does not working. Cisco 79x0, i dont know. I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco 7960's that would be capable of displaying which extension is being used and which extensions are not in use. Any suggestions would be appreciated. Thanks in advance, -Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Back On Busy?
Is there a functional reason why you'd use MeetMe here? I think probably the easiest way to accomplish this is to use an DeadAGI script which can be invoked via the 'h' extension in the context that would then perform the functionality you're looking for and if they get through it should just bridge the original caller back in.
On 9/26/05, Sherwood McGowan [EMAIL PROTECTED] wrote:
Anyone else out there have some thoughts? The customer wants to be able to control what can be redialed on busy, such as no international. I'm having my doubts as to whether or not this can be done. My idea seems like it would work, but after the customer hangs up, wouldn't the context stop processing?
Thanks,
SKM
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Damon EstepSent: Monday, September 26, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Call Back On Busy?
This may not apply to your situation, but many ATAs and SIP phones have this feature built in to the device.
We use Linksys/Sipura and auto redial and last call return work without any special setup.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Sherwood McGowanSent: Monday, September 26, 2005 7:45 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Back On Busy?
I know it's been touched on before, but no answers have been found to the best of my knowledge. I'm using a SIP only setup, with a sip provider giving PSTN and would like to see if anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly MeetMe?
I figure something like this, but want to get feedback
1. Get callers last dialed number, if international number, do not allow.
2. Playback a stuttertone to caller
3. Disconnect caller
4. Ring intended party check dial status. If busy, wait120 seconds and try again (do this for a total of 15 minutes)
5. If it's picked up, playback an announcement to the party and put them in a meetme conference
6. Ring the original caller and bridge them to the meetme conference.
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[Asterisk-Users] CAS Question
I have to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===Proprietary PBX(CAS)===IVR Server AG-E1 Regards _ Get free infected, boring, wrong, empty, or any other email for yourself. Go to --- http://www.mailchoose.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] didgium card in india
Capt MS wrote: thanks for the reply Is Digium card compatible with EPABX standards available in india , further how much does a card with three FXS and one FXO interface cost, Do u have any experience of implenting the same , I am in army what we lookin at is voice gateway to interface our PBX with the data network so that we have one underlying network to handle , any suggestions on how to implement in a cost effective manner. I am using Digium card in India (Trivandrum, Kerala) for a small call center application. What I did was to purchase the card in US, send it across to my friend in his US address and he brought it along when he came, but I guess this option is not applicable to you. 3 FXS and 1 FSO will cost some thing under Rs. 15,000, with out duty. See here for exact prices. http://store.yahoo.com/asteriskpbx/noname.html I tried it here with BSNL and a Siemens PBX, I am not receiving the callerid and it does not detect remote hangup. Pl mail me offline if you need further information. regards, raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
