[asterisk-users] Hardware Platform
We are in the process of building out www.dialaway4free.com, a free world wide calling service. I am writing RFQ's for hardware, since we are going to use asterisk as our call processor. I was wondering what is the best server platform to use that will support digium cards and handle sip termination for both clients and service providers. Also should I go with the open source of asterisk as compared to Asterisk for Business. Please let me know. I want this system to be stable as we will do a lot of proprietary programming for it to switch to the advertising component so I want to know what people think to handle the a call volume of at least 100,000 calls an hour. Some of my choices: Dell Gateway Gigabyte Ausus Please advise what type of processors and how much memory and hard drives, there will be no voicemail initially maybe it will be offered at a later time. Thanks Visit www.dialaway4free.com and register for a free account and be ready for our September 1st 2008 launch. Worldwide calling to and from any country! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cool New Website
Cool New Website For everyone to see! I think they are using a specially programmed version of Asterisk to do this. www.dialaway4free.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Support Question
I am thinking of building an Asterisk PBX, and had a question on a piece of hardware support. I want to include a 4 port PCI 10/100 Switch router card. For those not familiar it's a PCI card that acts as a switch. My question is would I be able to configure those 4 ports to support sip phones plugged in directly to the asterisk box instead of a switch. Thanks in advance ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Setup Question -- Please Help
I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines through the Cisco Call Manager. I want to set up a dial plan where 7 would grab the fx0 line for internal and the users would be able to place internal calls through the Cisco Call Manager. I envision people dialing 7 (4 digit extension.) This would call internally. I then envision setting up a calling plan where 7 would grab the trunk and 8 would grab an outside line from the Cisco Call Manager and then dial the 10 digit telephone number. 78xx. This would allow them to place external calls through the call manager. Is this something that would be feasible? Since the company is not looking to invest a lot in upgrading the Cisco yet they want to allow external sales reps to work from home. Would there be a way through Asterisk where I can then program the FX0 extension coming in from the Cisco Call Manager to ring into the Audiocodes and be dialed directly to an extension in the Asterisk server? Example - 1300---200 on the Asterisk. This would allow people calling the company to directly dial their sales people and be forwarded to the extension attached to the audiocodes. If this is feasible please let me know as I would like to propose this solution to the company. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GrandSTream 488/Asterisk
Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I read that the 488 has a FXO port on it, can I use the grandstream 488 to pass traffic to the pstn from Asterisk. I would use this at home to pass traffic into a foreign country's PSTN over the internet. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] choppy music on hold - only on PRI PSTN
Hello to all I do not know what is causing choppy music on hold when call comes in through E1 card (PRI).. but this channel info is somehow strange.. We use Alaw over PRI (and I think its format number 8), But why is WriteFormat at 2 ? Thanks! show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1136667936.0 Caller ID: 04573573 Caller ID Name: (N/A) DNID Digits: 349 State: Up (6) Rings: 1 NativeFormat: 72 WriteFormat: 2 ReadFormat: 8 1st File Descriptor: 14 Frames in: 3516 Frames out: 3352 Time to Hangup: 0 Elapsed Time: 0h1m10s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: OZ0800 Extension: s Priority: 7 Call Group: 0 Pickup Group: 0 Application: Queue Data: OZ0800|Tt|||300 Blocking in: ast_waitfor_nandfds ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ipVolution
Hi, Anybody have some experience and did some testing with ipVolution E1/T1 cards? goran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Error
I recently installed a 1 port Fxo card It detected the card when it was booting the Zaptel hardware was being detected upon bootup. I did a yum on Centos and then did a rebuild And then did an autoconfigure everything was working fine. Now when I reboot the zaptel is not coming on-line. Any suggestions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Product ID.
I am asking all the VOIP Gurus and any developers out there if a product exists and if not if anyone would want to help me develop such product. With the onslaught of new homes that are wired with networking capabilities. I was wondering if there is a product out there developed that can be used by Asterisk for intercom systems in homes, business or multi-dwelling buildings. I want to know if there is a system that you can install that will use SIP as the communication mechanism but install in every room and dial the extension of the rooms or do an extension that does a broadcast for all the intercoms. If this product exists can someone tell me who makes it and point me out to the websites. If not if someone is interested in developing such a product and cobranding it let me know. This unit would be an all in one system wall mounted in rooms that can be used inside or outside of entrance doors without a special intercom system. I believe that such a device would allow better marketing for Asterisk and VOIP systems to make their entrance in the residential field. This would allow builders to further push VOIP in their new dwellings. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum Calls handled
I have a big dilemma. I have a client who is looking for a big installation. I am looking at the digium product and have the following Questions. Difference between Asterisk and Asterisk Business Edition. My Client has 300 personal split between two office and wants to use one asterisk box to support those calls. He is going to have 3 PRIS coming in. Can I use the regular version of Asterisk compared to the Business Edition of Asterisk. How many simultaneous calls can Asterisk support compared to the Business Edition of Asterisk. Please help me out as I dont want to make the wrong recommendations. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] default user name and password for a2billing
What is the default username and password for [EMAIL PROTECTED] a2billing module. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Asterisk do This?
I have a client who is looking for the proposed solution and was wondering if any asterisk professionals know if this can be done by asterisk. Calling card platform. Users calling in through local access numbers, they dial local access numbers and make calls through the system to make affordable long distance lines. The lines would be coming to a PRI gateway probably MediaTrix or asterisks directly via a PRI card. They want the calling card platform to identify the users pin through Caller ID. Either if they call from home or they call phone. If they call from a 3rd party location to give them choice to enter their pin to be authorized by the system for them to make a outbound calling. These calls would be registered to their account and would be bill accordingly to the rates given to them. They want easy administration of this software, I saw A2Billing but I didnt see a part to identify the Pin through caller id. They want this software to be GUI driven and to be easy to administer. 2nd part they want is a VOIP Platform for VOIP ATAs for internet clients. They want to be able to attach ATA clients with DID numbers to they can make calls from their homes and receive incoming calls through this system. This part I know Asterisk can do, but I want to know if this is possible with the system they are looking to implement to have the complete package. They want the system to have a nice GUI like AMP to make the changes. If anyone knows how this can be done affordably with a small startup pilot system. Please let me know if this can be done it would be greatly appreciated. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP Installation
Has anyone had any success installing AMP 1.10 on a Asterisk 1.2.0. If so can anyone shed some light on how to install it? I am looking for an install or someone sort of script to run the installation and I can t see it. Any assistance would be appreciated. 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] Which is Better!
Which FXO gateway is better and has better sound quality. AudioCodes? Or Mediatrix. Thanks for your input ___ --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] Please Help with Zaptel
Can someone tell me what problem I am having with Zaptel on a Suse 10 distribution? 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\ 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/2.6.13-15-default/build make -C /lib/modules/2.6.13-15-default/build SUBDIRS=/root/zaptel-1.2.0 modules make[1]: Entering directory `/usr/src/linux-2.6.13-15-obj/i386/default' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.13-15-obj/i386/default' make: *** [linux26] Error 2 ___ --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] AMP installation
How do you install AMP? I downloaded it and tried to run make or install and it doesnt work. Is there some trick to this? Thank.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
[Asterisk-Users] Asterisk Compilation Error
l -lpthread -lncurses -lm -lresolv -lssl /usr/lib/gcc/i586-suse-linux/4.0.2/../../../../i586-suse-linux/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 Can someone tell me whats going on? ___ --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] Zaptel Error
Is this a bug in Zaptel configs. I have this message in Suse 10.0, and I had a similar message in Centos that the source files could not be found. Please shed some light on this. /lib/modules/2.6.13-15-default/build make -C /lib/modules/2.6.13-15-default/build SUBDIRS=/root/zaptel-1.2.0 modules make[1]: Entering directory `/usr/src/linux-2.6.13-15-obj/i386/default' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.13-15-obj/i386/default' make: *** [linux26] Error 2 asterisk:~/zaptel-1.2.0 # ___ --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
[Asterisk-Users] Zaptel Compile Error
First Thanks to all who worked hard to release 1.20! I installed asterisk with no problem and when it came to installing the zaptel drivers I am getting the following errors. Can anyone help me? The error message is: You do not appear to have the sources for the 2.6.9-22.0.1.EL kernel installed. make: *** [linux26] Error 1 I am installing it on a Cento 4.2 server. Can someone shed some light on this? ___ --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] [EMAIL PROTECTED] KDE or GNOME?
Do [EMAIL PROTECTED] have KDE or GNOME? How to start GUI? ___ --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 iptables rules
The simple solution was that I was missing: iptables -A INPUT -i eth0 -m state --state ESTABLISHED,RELATED -j ACCEPT Which caused replies to outgoing traffic to be stopped in the firewall... So problem wasnt really related to asterisk at all...doh - Original Message - From: Goran Tornqvist [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 28, 2005 9:52 AM Subject: Re: [Asterisk-Users] Asterisk iptables rules Hello, After further checking I found that when activating the firewall no traffic is allowed OUT from the box. Nameresolving, http, nothing accept ICMP works, even though I added: iptables -A OUTPUT -p all -j ACCEPT So I think its not related to asterisk at all, rather some iptables config problem... I'll see if I can fix that problem first...thats maybe the reason why it doesnt work. Thanks for your help anyway... Best Regards Goran - Original Message - From: Steve Davies [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 27, 2005 12:10 PM Subject: Re: [Asterisk-Users] Asterisk iptables rules I would suggest that you are missing something like: iptables -A INPUT -m state --state ESTABLISHED,RELATED -i eth0 -j ACCEPT This will mean that if a UDP packet is sent by * from sport:2345, dport:5060, then the response (sport:5060, dport:2345) will be allowed in, whereas at present that is not the case. I cannot say whether this type of packet will ever be sent, but I always include the rule for completeness. Alternatively, add a LOG rule, just before the DROP rule, and see what is being dropped... Regards, Steve On 10/27/05, Goran Tornqvist [EMAIL PROTECTED] wrote: One last check...won't ask again, promise :) Does someone know a solution to my problem below? Best Regards Goran - Original Message - From: Goran Tornqvist To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 10:33 AM Subject: Asterisk iptables rules Hello, I have trouble getting asterisk to work with my new firewall script (see below). I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules And then modified it to suit my needs. I use only SIP and the problem is that the calls get in to asterisk when the firewall is activated. But my agents/phones cant register or receive any calls. So all calls get stuck in queue on asterisk. So I believe Im missing some rule perhaps? Can anyone help me sort this out? Thanks... Best Regards Goran /etc/init.d/firewall == #IPTables firewall configuration for X export PATH=$PATH:/sbin case $1 in start) echo Starting iptables firewall... iptables --flush iptables --delete-chain iptables -A INPUT -p icmp -i eth0 -j ACCEPT # START OPEN PORTS #= #SSH (22) iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT #SAMBA: netbios (139) , microsoft-ds (445) iptables -A INPUT -p tcp -i eth0 --dport 139 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 445 -j ACCEPT #ASTERISK # SIP (UDP 5060) iptables -A INPUT -p tcp -m tcp -i eth0 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5060 -j ACCEPT # IAX2/IAX iptables -A INPUT -p udp -m udp -i eth0 --dport 4569 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp -i eth0 --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp -i eth0 --dport 2727 -j ACCEPT #END ASTERISK #MySQL (3306) iptables -A INPUT -p tcp -i eth0 --dport 3306 -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 3306 -j ACCEPT #SNMP (161) - Allow from cacti server iptables -A INPUT -p tcp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT #Ftp / Passive ports iptables -A INPUT -p tcp -i eth0 --dport 21 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 64785:64799 -j ACCEPT #Http / Web iptables -A INPUT -p tcp -i eth0 --dport 80 -j ACCEPT #Webmin (1) iptables -A INPUT -p tcp -i eth0 --dport 1 -j ACCEPT # END OPEN PORTS #= #Deny everything else iptables -A INPUT -p all -i eth0 -j DROP exit 0; ;; stop) echo Stopping iptables firewall... iptables --flush iptables --delete-chain exit 0; ;; *) echo Valid switches: firewall start , firewall stop; esac; ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk iptables rules
Hello, After further checking I found that when activating the firewall no traffic is allowed OUT from the box. Nameresolving, http, nothing accept ICMP works, even though I added: iptables -A OUTPUT -p all -j ACCEPT So I think its not related to asterisk at all, rather some iptables config problem... I'll see if I can fix that problem first...thats maybe the reason why it doesnt work. Thanks for your help anyway... Best Regards Goran - Original Message - From: Steve Davies [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 27, 2005 12:10 PM Subject: Re: [Asterisk-Users] Asterisk iptables rules I would suggest that you are missing something like: iptables -A INPUT -m state --state ESTABLISHED,RELATED -i eth0 -j ACCEPT This will mean that if a UDP packet is sent by * from sport:2345, dport:5060, then the response (sport:5060, dport:2345) will be allowed in, whereas at present that is not the case. I cannot say whether this type of packet will ever be sent, but I always include the rule for completeness. Alternatively, add a LOG rule, just before the DROP rule, and see what is being dropped... Regards, Steve On 10/27/05, Goran Tornqvist [EMAIL PROTECTED] wrote: One last check...won't ask again, promise :) Does someone know a solution to my problem below? Best Regards Goran - Original Message - From: Goran Tornqvist To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 10:33 AM Subject: Asterisk iptables rules Hello, I have trouble getting asterisk to work with my new firewall script (see below). I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules And then modified it to suit my needs. I use only SIP and the problem is that the calls get in to asterisk when the firewall is activated. But my agents/phones cant register or receive any calls. So all calls get stuck in queue on asterisk. So I believe Im missing some rule perhaps? Can anyone help me sort this out? Thanks... Best Regards Goran /etc/init.d/firewall == #IPTables firewall configuration for X export PATH=$PATH:/sbin case $1 in start) echo Starting iptables firewall... iptables --flush iptables --delete-chain iptables -A INPUT -p icmp -i eth0 -j ACCEPT # START OPEN PORTS #= #SSH (22) iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT #SAMBA: netbios (139) , microsoft-ds (445) iptables -A INPUT -p tcp -i eth0 --dport 139 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 445 -j ACCEPT #ASTERISK # SIP (UDP 5060) iptables -A INPUT -p tcp -m tcp -i eth0 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5060 -j ACCEPT # IAX2/IAX iptables -A INPUT -p udp -m udp -i eth0 --dport 4569 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp -i eth0 --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp -i eth0 --dport 2727 -j ACCEPT #END ASTERISK #MySQL (3306) iptables -A INPUT -p tcp -i eth0 --dport 3306 -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 3306 -j ACCEPT #SNMP (161) - Allow from cacti server iptables -A INPUT -p tcp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT #Ftp / Passive ports iptables -A INPUT -p tcp -i eth0 --dport 21 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 64785:64799 -j ACCEPT #Http / Web iptables -A INPUT -p tcp -i eth0 --dport 80 -j ACCEPT #Webmin (1) iptables -A INPUT -p tcp -i eth0 --dport 1 -j ACCEPT # END OPEN PORTS #= #Deny everything else iptables -A INPUT -p all -i eth0 -j DROP exit 0; ;; stop) echo Stopping iptables firewall... iptables --flush iptables --delete-chain exit 0; ;; *) echo Valid switches: firewall start , firewall stop; esac; ___ --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] Asterisk iptables rules
One last check...won't ask again, promise :) Does someone know a solution to my problem below? Best Regards Goran - Original Message - From: Goran Tornqvist To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 10:33 AM Subject: Asterisk iptables rules Hello, I have trouble getting asterisk to work with my new firewall script (see below). I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules And then modified it to suit my needs. I use only SIP and the problem is that the calls get in to asterisk when the firewall is activated. But my agents/phones cant register or receive any calls. So all callsget stuck in queue on asterisk. So I believe Im missing some rule perhaps? Can anyone help me sortthis out? Thanks... Best Regards Goran /etc/init.d/firewall == #IPTables firewall configuration for X export PATH=$PATH:/sbin case "$1" in start) echo "Starting iptables firewall..." iptables --flush iptables --delete-chain iptables -A INPUT -p icmp -i eth0 -j ACCEPT # START OPEN PORTS #= #SSH (22) iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT #SAMBA: netbios (139) , microsoft-ds (445) iptables -A INPUT -p tcp -i eth0 --dport 139 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 445 -j ACCEPT #ASTERISK # SIP (UDP 5060) iptables -A INPUT -p tcp -m tcp -i eth0 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5060 -j ACCEPT # IAX2/IAX iptables -A INPUT -p udp -m udp -i eth0 --dport 4569 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp -i eth0 --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp -i eth0 --dport 2727 -j ACCEPT #END ASTERISK #MySQL (3306) iptables -A INPUT -p tcp -i eth0 --dport 3306 -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 3306 -j ACCEPT #SNMP (161) - Allow from cacti server iptables -A INPUT -p tcp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT #Ftp / Passive ports iptables -A INPUT -p tcp -i eth0 --dport 21 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 64785:64799 -j ACCEPT #Http / Web iptables -A INPUT -p tcp -i eth0 --dport 80 -j ACCEPT #Webmin (1) iptables -A INPUT -p tcp -i eth0 --dport 1 -j ACCEPT # END OPEN PORTS #= #Deny everything else iptables -A INPUT -p all -i eth0 -j DROP exit 0; ;; stop) echo "Stopping iptables firewall..." iptables --flush iptables --delete-chain exit 0; ;; *) echo "Valid switches: firewall start , firewall stop"; esac; ___ --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 iptables rules
Hello, I have trouble getting asterisk to work with my new firewall script (see below). I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules And then modified it to suit my needs. I use only SIP and the problem is that the calls get in to asterisk when the firewall is activated. But my agents/phones cant register or receive any calls. So all callsget stuck in queue on asterisk. So I believe Im missing some rule perhaps? Can anyone help me sortthis out? Thanks... Best Regards Goran /etc/init.d/firewall == #IPTables firewall configuration for X export PATH=$PATH:/sbin case "$1" in start) echo "Starting iptables firewall..." iptables --flush iptables --delete-chain iptables -A INPUT -p icmp -i eth0 -j ACCEPT # START OPEN PORTS #= #SSH (22) iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT #SAMBA: netbios (139) , microsoft-ds (445) iptables -A INPUT -p tcp -i eth0 --dport 139 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 445 -j ACCEPT #ASTERISK # SIP (UDP 5060) iptables -A INPUT -p tcp -m tcp -i eth0 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5060 -j ACCEPT # IAX2/IAX iptables -A INPUT -p udp -m udp -i eth0 --dport 4569 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp -i eth0 --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp -i eth0 --dport 2727 -j ACCEPT #END ASTERISK #MySQL (3306) iptables -A INPUT -p tcp -i eth0 --dport 3306 -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 3306 -j ACCEPT #SNMP (161) - Allow from cacti server iptables -A INPUT -p tcp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT #Ftp / Passive ports iptables -A INPUT -p tcp -i eth0 --dport 21 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 64785:64799 -j ACCEPT #Http / Web iptables -A INPUT -p tcp -i eth0 --dport 80 -j ACCEPT #Webmin (1) iptables -A INPUT -p tcp -i eth0 --dport 1 -j ACCEPT # END OPEN PORTS #= #Deny everything else iptables -A INPUT -p all -i eth0 -j DROP exit 0; ;; stop) echo "Stopping iptables firewall..." iptables --flush iptables --delete-chain exit 0; ;; *) echo "Valid switches: firewall start , firewall stop"; esac; ___ --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] How to configure two Asterisk servers for onecall center
On Fri, 2005-10-21 at 09:39 -0700, Tielin Xu wrote: Hi All: I have a situation to be resolved. Assume that one location call center with 150 agents. I have two asterisk servers to serve those 150 sip phones. The servers are connected to PSTN as 4 T1/PRI for each. My question is why do you have about 150% the agents to the line capacity? Even with pauses and all do you expect that the 96 (or less in the case of pri) lines to be in use at all times? Predictive dialing ? ___ --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] Voicemail as an email attachement
Small.. just app_voicemail.c and a sendEmail script... You can download it from here: app_voicemail.c http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=9 and sendEmail http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=10 sendEmail is most important.. code change is really small in app_voicemail.. but here it is.. 1. install sendEmail 2. Edit app_voicemail.c : You will need to change app_voicemail.c to suit your needs.. Go to line 1035 (or find goran.skular) and: Change [EMAIL PROTECTED] to from address you want to show up Mail.slsolucije.hr:25 change to your mail.server.xxx:smtp Password_here is place for your password.. Go to line 1130 also (or find next appereance of goran.skular) and to the same again. That's all in short. Have a nice day. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 19, 2005 4:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail as an email attachement Yes. I am interested. I will make provisions for the upload. How big are the files? Thanks BEN Goran Skular wrote: I changed my app_voicemail.c to work not with sendmail but with sendEmail that connects to any SMTP and sends email with attachment... It's dirty, but it works. If you are interested I can upload app_voicemail.c and sendEmail package somewhere.. I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can anyone point me to the next step to setup the attachment of voicemail messages to an email? 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 ___ --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] Voicemail as an email attachement
On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote: I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can anyone point me to the next step to setup the attachment of voicemail messages to an email? Set up a sendmail. Or basically: an MTA. Any linux distro comes with at least one (postfix seems to be the preffered choice nowadays). Which one do you use? There are a bunch of programs that provide /usr/sbin/sendmail but don't spool the result. Check msmtp, ssmtp, masqmail and nullmailer. There are probably others. The downside is that messages that have, for some reason, not been delivered in the first shot (e.g: due to some transient network error) will be dropped rather than queued. I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. ___ --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 in Croatia - Zagreb
Hello, I was wondering how many people from Croatia are using and playing with Asterisk. Recently I had a contact with one user and I am very glad. It will be really nice to organize a Croatian Asterisk community and on that way we are organizing a little gathering. It does not matters how much experience you have, everthing you need is some interest in Asterisk. Beside my last contact I know that croatian wifi community ZG Wireless is using Asterisk also. So, Everyone of you, located in Croatia, please contact me here or on email. For the purpose of collecting as much people, gathering is to be expected next month (around 19th) Send me an e-mail or even register on www.migo-systems.com. Further info will be available later. Looking forward for it, Goran Skular www.slsolucije.hr ___ --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] Voicemail as an email attachement
On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote: I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. Sending is never a problem. Recieving is a problem when you're on a dynamic address. You can tell your MTA to do just that. e.g, on postfix, in /etc/postfix/main.cf: # assuming a well-behaved setup relayhost = the.isp.domain # and if not: relayhost = [smtp.the.isp.domain] BTW: one option you have with a decent mailer is not to write the email address in voicemail.conf, but rather, to write there for each box the email vmbox-vmbox, and use the MTA's aliases to map them to emails. Either using a plain text /etc/aliases, or using any other database (ldap, mysql, whatever). If relaying is enabled and accepted on remote side... and nowdays is hard to enable relaying with those spammers around.. I tried something with this relaying, but without success, so I changed app_voicemail in order to send mail with SMTP and sendEmail script. Can you tell me how to accept relaying on server, but to limit it to allowable IP address (which is in this case dynamic ip..). That will help me a lot :) ___ --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 in Croatia - Zagreb
Hello, I'm there with you, dude, haven't talked to you in some 5-6 years? :) I know a couple of people that are working with Asterisk... Cheers, Vedran. Nice surprise ! :) Ok, you're the first participant along with me on this small gathering. I sent you email, and let's ring on those guys you know. I hope that we will find some people out there for a nice gathering on that subject (and subjects involved in our past 5-6 years you mentioned :) ) See you, Goran ___ --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] New ISDN architecture available for asterisk
Hi to all, sorry for crossposting the -dev and -user lists, but I think this could be quite interesting news for EuroISDN people, expecially BRI owners. A new ISDN architecture, called vISDN, has been developed to fully support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and HFC-8S (with HFC-E1 and HFC-S USB support coming soon). vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc... but has been designed from scratch to be a standard compliant EuroISDN implementation plus a channel crossconnector, plus protocol analisys support thru Ethereal, plus a ppp terminator, plus other stuff :) Very, very nice.. I am looking forward for test it. Further, I hope that ecgo cancelation will be implemented also in near future, as it is very important in most cases. Are there maybe some HFC (both BRI and PRI) boards with hw echo cans, or they are all passive? For small Euro BRI installations we are using at this moment HFC with bristuff. But where E1 is involved, we are trying now to avoid E1 cards without HW echo cans integrated. At this point we are considering between Sangoma and Digium with hw cans... but who knows what HFC boards would bring. Beronet and Junghanns are here to be observed.. Kind regards, Goran ___ --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] Voicemail as an email attachement
Always download programs directly from the homepage or from another reliable source. Don't just grab programs and scripts from everywhere. But why not just set mailcmd in voicemail.conf? Also, quoting the homepage: Why not use sendmail? Sendmail is a large and complex mail server. Installing this kind of mail software on servers (unless it's a mail server) is more of a security risk than its worth. Not if it only listens on localhost or doesn't listen at all. The codebases of sendmail is indeed known to be a source of many security breaches, but exim, postfix and qmail are not so. Most distros come with either postfix or exim by default nowadays. Not to mention it can be a real pain messing with configuration files and such. Systems need another simpler way to send email from the command prompt, and sendEmail provides this functionality. Its a simple, direct way to send email without the overhead of other conventional email software. Most of the pain is caused due to management of messages in the queue. Other types of pain are due to messages routing. Routing issues can be easily solved by sending basically all mail to a remote host (excpt, maybe, some system messages). However, if the system is disconnected from the net for a while what will you do? lose all voicemail messages? (and get just ugly warnings in the logs as a reminder) Also note that there are quite a few programs that could use a sendmail-compatible interface. cron sends its output using mail. So are many other programs. If you don't provide a sendmail-compatible interface (even if it one that does not queue, something like nullmailer) you'll have to reconfigure other parts of your system as well. And worst of all: you won't be able to send mail with mutt. The horors! I completely agree! This is only a work-around.. there are much better methods involved with sendmail which is really powerfull and thus really complicated to configure. The most difficult part is not on * server side, but on relaying server side which must be configured to allow relays only from authorized sites.. I had no success with that, even with some keys and similar solutions which I tried, so I gave up and start using sendEmail. But, I will for sure migrate to sendmail when time for that comes, and I strongly suggest it to everyone. ___ --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 in Croatia - Zagreb
At this moment we are counting 4 possible participants. (Appoligies for those who are not from Croatia for using this list, but this list has a lot of subscribers including from Croatia) We are waiting for others to join us. Feel free to respond here or on my e-mail. Thanks! Hello, I was wondering how many people from Croatia are using and playing with Asterisk. Recently I had a contact with one user and I am very glad. It will be really nice to organize a Croatian Asterisk community and on that way we are organizing a little gathering. It does not matters how much experience you have, everthing you need is some interest in Asterisk. Beside my last contact I know that croatian wifi community ZG Wireless is using Asterisk also. So, Everyone of you, located in Croatia, please contact me here or on email. For the purpose of collecting as much people, gathering is to be expected next month (around 19th) Send me an e-mail or even register on www.migo-systems.com. Further info will be available later. Looking forward for it, Goran Skular www.slsolucije.hr ___ --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] Voicemail as an email attachement
I changed my app_voicemail.c to work not with sendmail but with sendEmail that connects to any SMTP and sends email with attachment... It's dirty, but it works. If you are interested I can upload app_voicemail.c and sendEmail package somewhere.. I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can anyone point me to the next step to setup the attachment of voicemail messages to an email? 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 ___ --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] Middle Ground between POTS and T1?
Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of our telcos (DT T-com) we can get PRA in 10 increments: 10B, 20B and 30B We have a partial T1 (5B + D, iirc) from Allstream - there may be a provider in your area that does something similar. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 12:48 PM, Matthew T. O'Connor wrote: I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but smaller. Thanks, Matthew ___ --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 show peers
They do not have NAT option.. and they do not have qualify... Hi, I have 3 SIP extensions, setup as follows: # Device Location options 200 Sipura local 210 Sipura remote nat=yes qualify=yes 310 eyebeam remote nat=yes qualify=yes This is the result of sip show peers: Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards ___ --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] SS7 with Asterisk
Name of the company is MULTI-line GmbH You can contact mr. Zlatko Medibach at +43 1 78932320 or on cellular +43 676 3220262. Mail: [EMAIL PROTECTED] Their HQ is in Wien.. I can not help you with the details, I just know that they implemented SS7 on * for some telcos there. Goran -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Johann Steinwendtner Sent: Tuesday, October 11, 2005 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SS7 with Asterisk Goran, which company ist this ? Do they use the www.ss7box.com approach ? Thanks and best regards Hans Goran Skular schrieb: anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... I can point you to one company in Austria. They deployed SS7 on Asterisk, but not with Digium cards for one smaller telco. ___ --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] arcaplex / horizon isdn and analog multiplex
Has anybody tried something like this: http://www.arca-technologies.com/datasheets/arcaplexhorizon.pdf It will be interesting to have ability to make systems like: SCENARIO 1 (2 incoming BRI lines and 12 analog extensions with ability to connect additional isdn devices to s0 buses): 1 card with 8 BRI (from Junghanns or Beronet or someone else) ports (2 of them configured in TE mode and 6 of them in NT mode) 1 something that will convert e.g. 6 BRI to 12 analog FXS ports for analog telephony equipment.. Or SCENARIO 2 (1 incoming PRI E1 or T1 and 32 or more analog extensions) 2 Digium/Sangoma/Eicon whatsoever T1 or E1 cards (1 to telco, 1 to something like this arcaplexhorizon) 1 Arcaplexhorizon ISDN and analog multiplexor with 32 analog ports (PO21/32A) Maybe this Arcaplex can be used for 32 analog ports connected to Asterisk with 1E1/T1 card Some thoutghts ? Goran ___ --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] Outgoing call: hangup after answer
For your information.. if someone get in the same trouble.. problem is solved, but not with the software We just changed our BRI NT device with a different one.. from now on it works very well We had Elcon NT1+2a/b and now it is replaced with Santis ISDN NT1+2ab Here is pri debug: -- Making new call for cr 143 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=22 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 81] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 ChanSel: B1 channel ] [6c 05 21 80 32 30 30] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '200' ] [70 01 c1] Called Number (len= 3) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '' ] -- Called g1/ Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 143/0x8F) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: INFORMATION (123) [70 02 c1 30] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0' ] Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: INFORMATION (123) [70 02 c1 39] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0' ] Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 143/0x8F) (Terminator) Message type: CALL PROCEEDING (2) -- Zap/1-1 is making progress passing it to SIP/200-7b76 Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 143/0x8F) (Terminator) Message type: ALERTING (1) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) n Zap/1-1 is ringing NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Hi, When we make an outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup after answer. But when we place a full number before dialing everything is ok. Any help appriciated!! Thanks here is info with debug: [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [1e 02 84 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 30 (cs0, Progress Indicator) -- Zap/1-1 is ringing, hanging up. NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 64/0x40) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] ___ --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:
[Asterisk-Users] sound very loud (saturated) through IAX2 and SIP
I have very loud sound through IAX2 and SIP channels, even very saturated in some moments. Why? How to change sound level (on IAX2 and SIP channels)? ___ --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] Open Source Content Management System - Joomla
Our Web is based on Mambo portal software and it is connected with our Asterisk installation. We wrote our own CDR rating engine and modules for Mambo. Also, you can register for VoIP termination services inside mambo.. we wrote one component and couple of modules. So, when user create an account on mambo, asterisk account is also created automaticly (if choosed... cron script every 10 minutes) ... CDR rating is simple, but it works.. there is no fancy things in rating like tariffs or similar... (at this moment) You can check on www.slsolucije.hr .. it is on Croatian.. but.. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Monday, October 10, 2005 11:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Open Source Content Management System - Joomla And how exactly is Asterisk relevant to a CMS? could you give a more specific example? This is relevant where Administrative users wanted to manage their Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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] SS7 with Asterisk
anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... I can point you to one company in Austria. They deployed SS7 on Asterisk, but not with Digium cards for one smaller telco. ___ --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] Outgoing quality
We were using ilbc at first.. but it shows that it really needs a lot of time for transcoding.. (CLI: show translation) resulting with hearable delays. So, where bandwidth is an issue, try to use g729. We are also using gsm, it seems that it works very well. OK .. what about ilbc ... could it be a decent choice?? -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] conto di Mojo with Horan Company, LLC Inviato: lunedì 10 ottobre 2005 22.00 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Outgoing quality Are you calling from a soft- or hardphone on a network with a high amount of latency? If your (for example) SIP phone can't deliver voice packets to asterisk in time for asterisk to put them where they belong in the Zap channel, things like this might happen. Usually the interruptions could be described as clicks or crackles. In this case, you could reduce the network traffic by utilizing a codec with a smaller bandwidth usage, like g729 or gsm if your phone supports it. Fabrizio Mazzoni wrote: ___ --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] [Fwd: Libpri/chan_zap problems?]
What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer: Unable to forward voice Does the same thing happens even when you're not calling cellular number VIP (I assume you are in Croatia, calling VIPnet) i.e. some fixed line number ? And what connection do you use, BRI (bristuff, capi), PRI, some FXO,... You can reach me at 01/4573573. I'll be glad to hear you if my assumptions (on Croatia thing) were right... ___ --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] Outgoing call: hangup after answer
Hi,
When we make an
outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup
after answer. But when we place a full number before dialing everything is ok.
Any help appriciated!! Thanks
here is info with
debug:
== Primary
D-Channel on span 1 up -- Executing Dial("SIP/200-164c",
"zap/g1/|100|tc") in new stack-- Making new call for cr
192 -- Requested transfer capability: 0x00 -
SPEECH Protocol Discriminator: Q.931 (8) len=22 Call Ref:
len= 1 (reference 64/0x40) (Originator) Message type: SETUP (5)
[04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std:
0 Info transfer capability: Speech
(0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35) [18 01 81]
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred
Dchan:
0
ChanSel: B1
channel
] [6c 05 21 80 32 30 30] Calling Number (len= 7) [ Ext: 0
TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163)
(1)
Presentation: Presentation permitted, user number not screened (0) '200'
] [70 01 c1] Called Number (len= 3) [ Ext: 1 TON:
Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
'' ] -- Called g1/ Protocol Discriminator: Q.931
(8) len=11 Call Ref: len= 1 (reference 192/0xC0)
(Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 01
89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive Dchan:
0
ChanSel: B1
channel
] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network
serving the local user
(2)
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ]-- Processing IE 24 (cs0, Channel Identification)--
Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931
(8) len=8 Call Ref: len= 1 (reference 64/0x40)
(Originator) Message type: INFORMATION (123) [70 02 c1
39] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number
(4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ]
Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1
(reference 64/0x40) (Originator) Message type: INFORMATION (123)
[70 02 c1 35] Called Number (len= 4) [ Ext: 1 TON: Subscriber
Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5'
] Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1
(reference 192/0xC0) (Terminator) Message type: CALL PROCEEDING
(2) -- Zap/1-1 is making progress passing it to
SIP/200-164c Protocol Discriminator: Q.931 (8) len=12 Call
Ref: len= 1 (reference 192/0xC0) (Terminator) Message type: ALERTING
(1) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network
serving the remote user
(4)
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ] [1e 02 84 82] Progress Indicator (len= 4) [
Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public
network serving the remote user
(4)
Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ]--
Processing IE 30 (cs0, Progress Indicator)-- Processing IE 30 (cs0, Progress
Indicator)
-- Zap/1-1 is ringing, hanging
up.
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered,
peerstate Call Received Protocol Discriminator: Q.931 (8)
len=8 Call Ref: len= 1 (reference 64/0x40) (Originator) Message
type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext:
1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private
network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
] -- Hungup 'Zap/1-1' Protocol Discriminator:
Q.931 (8) len=4 Call Ref: len= 1 (reference 192/0xC0)
(Terminator) Message type: RELEASE (77)NEW_HANGUP DEBUG: Calling
q931_hangup, ourstate Null, peerstate Release Request Protocol
Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference
64/0x40) (Originator) Message type: RELEASE COMPLETE (90) [08 02
81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate
Null -- Executing Macro("SIP/200-164c", "hangupcall") in
new stack -- Executing ResetCDR("SIP/200-164c", "w") in
new stack -- Executing NoCDR("SIP/200-164c", "") in new
stack -- Executing Wait("SIP/200-164c", "5") in new
stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/200-164c' in macro 'hangupcall' == Spawn extension
(from-internal, h, 1) exited non-zero on
'SIP/200-164c'
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[Asterisk-Users] overlap zaphfc - dialtone
Hello all, I have a problem with overlap dialing and don't know how to get rid of it. My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten = 222,1,Dial(zap/g1,100,tc) and channel got hangup every time. So I even saw bug 4913 http://bugs2.digium.com/view.php?id=4913nbn=1 and bug 4771. But that wasn't my problem my problem is that I didn't included / after g1.. So, I changed that and now my extension look like: exten = 222,1,Dial(zap/g1/,100,tc) This solved the problem with line being hangup-ed like in bug 4913, and I am getting the telco dialtone. So, when dialing 222 I get: A12*CLI -- Executing Dial(SIP/200-dd52, zap/g1/|100|tc) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/ After that, I can even dial from this dialtone, but when called party rings I get the following message and auto hangup: -- Zap/1-1 is making progress passing it to SIP/200-dd52 -- Zap/1-1 is ringing, hanging up. -- Hungup 'Zap/1-1' The called phone stops ringing, Zap channel hangs up, And SIP phone is still on the air without anybody. Thank You all for help, Goran ___ --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] overlap zaphfc - dialtone
My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten = 222,1,Dial(zap/g1,100,tc) and channel got hangup every time.. So I even saw bug 4913 http://bugs2.digium.com/view.php?id=4913nbn=1 and bug 4771.. But that wasn't my problem. my problem is that I didn't included / after g1.. So, I changed that and now my extension look like: exten = 222,1,Dial(zap/g1/,100,tc) This solved the problem with line being hangup-ed like in bug 4913, and I am getting the telco dialtone. So, when dialing 222 I get: -- Executing Dial(SIP/200-dd52, zap/g1/|100|tc) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/ -- Zap/1-1 is making progress passing it to SIP/200-dd52 -- Zap/1-1 is ringing, hanging up. -- Hungup 'Zap/1-1' It seems that when it detects Ringing, Asterisk executes Hangup in the same time But why? ___ --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] Differ between private and out of area?
usecallerid=yes hidecallerid=no callerid=asreceived usecallingpres=yes callwaiting=no callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=10 context=pstn rxgain=8.15 txgain=2.0 signalling=fxs_ks channel = 1 - Original Message - From: Rich Adamson [EMAIL PROTECTED] Paste the section from zapata.conf that handles the x101p. ___ --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] Differ between private and out of area?
Yes, I know that, but, how to distinguish between them at incoming call? - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: pon 19. sep 2005 16:22 Subject: Re: [Asterisk-Users] Differ between private and out of area? A private call is a call that someone has specifically blocked. An out of area or unknown call is simply a call that the caller-id did not come through on correctly, for some reason. On 9/18/05, Goran Dj. [EMAIL PROTECTED] wrote: Is there any method to make difference between Hidden (Private) and unknown (Out of area) incoming calls on ZAP/x101p? I want to block any hidden call, and to allow unknow calls, but ZAP channel (X101P) always delivering empty CALLERID= in both cases. ___ --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] Differ between private and out of area?
I have CID identificator connected in parallel with X101P/Asterisk, and it displays Private for hidden calls, and Out of area for calls from rural areas (with old phone systems). But Asterisk always deliver CALLERIDNUM=, CALLERIDNAME= in both cases. ___ --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] Looking for firmware for Cisco 12sp+ and 30VIP
Which firmware do you need? I have couple of them, but I don't know is it legal to share them over Internet :-)) - Original Message - From: Stern, Craig To: asterisk-users@lists.digium.com I have been looking for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any help in locating would be much appreciated. 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] Differ between private and out of area?
Is there any method to make difference between Hidden (Private) and unknown (Out of area) incoming calls on ZAP/x101p? I want to block any hidden call, and to allow unknow calls, but ZAP channel (X101P) always delivering empty CALLERID= in both cases. ___ --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] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is! ___ --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] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
No, no, it's more SIMPLE than that. Try this: [incoming] exten = s,1,setcallerid(NAMENUMBER) exten = s,2,dial() That's all (after couple of hours of investigation). If call origin from SIP, i see NUMBER on my phone, if call origin from PSTN, i see NAME on my phone. - Original Message - From: Shaun Ewing [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: pon 19. sep 2005 2:04 Subject: Re: [Asterisk-Users] sometimes CIDNUM shows, sometimes CIDNAME??? Why,why, why, why? On 9/19/05, Goran Dj. [EMAIL PROTECTED] wrote: Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is! Have you considered the possibility that your SIP provider may not be sending you the caller id name? CNAM looksup do cost money, and it's probably the exception rather than the norm to find a VoIP provider that will deliver it. ___ --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] X101P ringing too long !
Pause between successive incoming rings on my phone line is 4 sec, so when x101p do not receive next ring signal after 4.5 sec, call should be consider as ended. But, if caller hang-up, call is ended (Hungup 'Zap/1-1) exactly 8 sec after last ring, and my voip phone continues to ringing during that time (that's bad). I want to cut that time to 4.5 sec. How to do that? I tried to change in zapata.h some lines: #define ZT_DEFAULT_RINGTIME 500 #define ZT_LOOPCODE_TIME 3000 #define ZT_RINGOFFTIME 2000 but with no effects. Call is still ended 8 sec after last ring. ___ --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] Why ZAPATA inserting pause before last digit, during dialing? GRRRRR....
I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? I found what to change: digits.h line 23 from #define DEFAULT_DTMF_LENGTH 100 * 8 to #define DEFAULT_DTMF_LENGTH 50 * 8 and my dialling is now much faster. But, I have new question: Before last digit, there is always inserted pause (500ms) or maybe two (1000ms). I don't use pause anywhere in my dial-plan, so, why is inserted and dialled? To test is that really a pause or something else, I changed line 25 from #define PAUSE_LENGTH 500 * 8 to #define PAUSE_LENGTH 2000 * 8 and, guess what, now I have 4sec pause before last digit is played. How to get rid of it? I can maybe #define PAUSE_LENGTH 0 * 8 but that this is very dirty solution. ___ --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] How to speed-up INCOMING-RINGING-ENDED detection on X101P/zapata?
Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? I measured, event -- Hungup 'Zap/1-1' is shown exactly 8 sec after last detected ring (on X101P), and my voip phone continues to ringing during that time (that's bad). I want to cut that time to 4.5 sec. How to do that? I tried to change in zapata.h some lines: #define ZT_DEFAULT_RINGTIME 500 #define ZT_LOOPCODE_TIME 3000 #define ZT_RINGOFFTIME 2000 but with no effects. Hungup is still shown 8 sec after last ring. ___ --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] How to shorten ringing stop detection onX101Pclone?
This working only when zap answer call. But, if zap don't answer (ringing), and (outside) caller hangup, then there is no busy tone. By the way, do you know some voip provider in Paris with Direct Inward Dial numbers? Where can I found best information about prices of France Telecom (PRI od BRI ISDN/RNIS, tarrifs, etc...) - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: cet 1. sep 2005 7:13 Subject: RE : [Asterisk-Users] How to shorten ringing stop detection onX101Pclone? Hello Goran, Modify your /etc/asterisk/zapata.conf like this : busydetect=yes busycount=3 And, of course, you must have chosen your correct country for ringing mode in your /etc/zaptel.conf file : loadzone=fr defaultzone=fr I am in France :-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Goran Dj. Envoyé : jeudi 1 septembre 2005 02:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] How to shorten ringing stop detection on X101Pclone? When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? ___ --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] How to speed-up dialnig with X101P clone modem?
I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? (On usual modems, I can type ATS11=50 to get tone dialing much faster (50ms instead of default 90ms). After that, I can write configuration to nvram (ATW) to be permanent) ___ --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] How to shorten ringing stop detection on X101P clone?
When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? ___ --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] Detect Dialtone
Dialtone detection should be an option in .conf for zap channel, i agree with that. Are you trying to play with the case where you have an analog phone bridged on your fxo line, and detect the lack of dialtone when someone is using that analog phone? Belive or not, but at some places on the world are still in use some old (non-digital) ATC-es which do now provide dial-tone instantly. For example, when ATC ARF-102 is very congested with outgoing calls, you must wait some (unknown) time to get dialtone (10sec, 1min, 5min...) ___ --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] chan_capi on slackware10? cannot compile :-( why?
I'm trying to compile chan_capi-0.5.4 on Slackware 10, but I have bunch of errors. (By the way, can I use chan_capi for ISDN card with winbond w6692cf chipset?) I'm not a linux expert, still :-) Before compiling, when I type modprobe capi to load capi module, and then lsmod, i get list of modules: capi6208 0 kernelcapi 30496 1 [capi] capiutil 22272 0 [kernelcapi] uhci 2 0 (unused) usbcore59308 1 [uhci] hisax 448240 0 (unused) isdn 116684 0 [hisax] slhc4976 0 [isdn] wcfxo 8384 2 zaptel176992 8 [wcfxo] ide-scsi9328 0 ne 6672 1 83906000 0 [ne] crc32 2880 0 [8390] isa-pnp30736 0 [hisax ne] So, where is a problem? Should I compile kernel with capi as a part of a kernel, not as a module? How to do that? Errors when I try to compile chan_capi: [EMAIL PROTECTED]:#make ./create_config.sh /usr/include Checking Asterisk version... * no 'struct ast_channel_tech', using old pvt * ast_dsp_process() without 'needlock' * no 'struct ast_callerid' * found 'struct timeval delivery' * no 'transfercapability' * no 'ast_config_load' config.h complete. gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/i nclude -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DASTERISKVERSION=\\ -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:49:20: capi20.h: No such file or directory In file included from chan_capi.c:52: chan_capi_app.h:28: error: parse error before get_ast_capi_MessageNumber chan_capi_app.h:28: warning: type defaults to `int' in declaration of `get_ast_capi_MessageNumber' chan_capi_app.h:28: warning: data definition has no type or storage class chan_capi_app.h:34: error: parse error before _capi_put_cmsg chan_capi_app.h:34: error: parse error before '*' token chan_capi_app.h:34: warning: type defaults to `int' in declaration of `_capi_put_cmsg' chan_capi_app.h:34: warning: data definition has no type or storage class In file included from chan_capi.c:53: chan_capi_pvt.h:133: error: parse error before _cword . . . chan_capi.c:2834: error: invalid lvalue in assignment chan_capi.c:2835: error: invalid lvalue in assignment chan_capi.c:2837: error: invalid lvalue in assignment chan_capi.c:2844: error: `error' undeclared (first use in this function) chan_capi.c:2845: error: `CMSG2' undeclared (first use in this function) chan_capi.c:2847: warning: implicit declaration of function `IS_FACILITY_CONF' chan_capi.c:2875: error: subscripted value is neither array nor pointer chan_capi.c:2877: error: subscripted value is neither array nor pointer chan_capi.c:2882: error: subscripted value is neither array nor pointer chan_capi.c:2886: error: subscripted value is neither array nor pointer chan_capi.c:2890: error: subscripted value is neither array nor pointer chan_capi.c:2894: error: subscripted value is neither array nor pointer chan_capi.c:2898: error: subscripted value is neither array nor pointer chan_capi.c:2902: error: subscripted value is neither array nor pointer chan_capi.c:2906: error: subscripted value is neither array nor pointer chan_capi.c:2910: error: subscripted value is neither array nor pointer chan_capi.c:2914: error: subscripted value is neither array nor pointer chan_capi.c:2918: error: subscripted value is neither array nor pointer chan_capi.c:2922: error: subscripted value is neither array nor pointer chan_capi.c: In function `load_module': chan_capi.c:3088: warning: implicit declaration of function `capi20_isinstalled' chan_capi.c:3094: warning: implicit declaration of function `capi20_register' chan_capi.c:3104: warning: implicit declaration of function `capi20_get_profile' chan_capi.c: In function `unload_module': chan_capi.c:3301: warning: implicit declaration of function `capi20_release' make: *** [chan_capi.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?
capi6208 0 kernelcapi 30496 1 [capi] capiutil 22272 0 [kernelcapi] uhci 2 0 (unused) usbcore59308 1 [uhci] hisax 448240 0 (unused) isdn 116684 0 [hisax] slhc4976 0 [isdn] wcfxo 8384 2 zaptel176992 8 [wcfxo] ide-scsi9328 0 ne 6672 1 83906000 0 [ne] crc32 2880 0 [8390] isa-pnp30736 0 [hisax ne] So, where is a problem? Should I compile kernel with capi as a part of a kernel, not as a module? How to do that? It's okay to use it as modules. But the cards supported by HiSax do not provide CAPI interface. I don't know the status of mISDN, but that would be the driver supporting CAPI. Hmmm? I don't know what hisax doing here (and even what is that). My ISDN card (winbond w6692cf chip) isn't in computer, I will put it there when I successfully complile chan_capi. What modules do I need? Only capi(kernelcapicaputil) and chan_capi? You don't have libcapi20 (or the development package of it) installed. Yes, but I dont have it on my Slackware10 CD'es. I don't have libcapi, or isdn4... or anything with isdn or capi in their name. Where to find libcapi20 (od devel...) for slackware? Goran ___ Asterisk-Users mailing list Asterisk-Users@lists.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] chan_capi, cannot open /dev/capi20, no cards configured in /etc/capi.conf
wget ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-08-21.tar.bz2 tar xvjf isdn4k-utils-CVS-2005-08-21.tar.bz2 cd isdn4k* cd capi20 ./configure make make install that's all Sergio Ok. Thanks. It's working, and I compiled successfully chan_capi-0.5.3 (because 0.5.4 producing some error). But, now I cannot start chan_capi.so: WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI disabled! from tty: capiinit ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or directory (2) capiinfo capi not installed - No such file or directory (2) capiinit show ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2) ERROR: no cards configured in /etc/capi.conf So, whats happening? What is responsible for making /dev/capi20, and how to make /etc/capi.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
Re: [Asterisk-Users] chan_capi, modprobe: Can't locate module capifs, ERROR: fopen(/etc/capi.conf, r)
But, now I cannot start chan_capi.so: WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI disabled! from tty: capiinit ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or directory (2) capiinfo capi not installed - No such file or directory (2) capiinit show ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2) ERROR: no cards configured in /etc/capi.conf I resolved missing /dev/capi20 with shell script makedev-capi.sh but, now, when starting capiinit: modprobe: Can't locate module capifs modprobe: Can't locate module capifs WARNING: filesystem capifs not available ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2) ___ --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] Ast.1.0.9 (only) strange problem with IAX and DDNS
Asterisk 1.0.9: IAX2 registration timeout! --- I have 2 locations with ADSL lines, both with dynamic IP (+ dynamic DNS). On location 1 = Asterisk 1.0.RC2 / Slackware 10 On location 2 = Asterisk 1.0.9 / Slackware 10 They are on private network and connected via IAX2 through NAT(win2000server), and registering to DDNS name of each other. I know that Asterisk is not very smart on handling DNS, so when remote ADSL change IP address, I must reload IAX on local Asterisk to (resolve new address and) continue registering itself to remote Asterisk. But, here start problem: Asterisk 1.0.9 (location 2), sometimes (very often) when LOCAL ip address is changed, can't anymore register himself to remote Asterisk 1.0.RC2 which by the way DIDN'T change it's IP address! Remote Asterisk (unchanged IP) also can't register himself to local Asterisk (changed IP) even when I do reload of IAX (on remote Asterisk). That problem cannot be resolved with unload/load IAX2, or stop/start Asterisk. Only reboot of local Slackware (location 2, unchanged IP, Asterisk 1.0.9) helping, and after reboot everything working well (till some of next IP address changing). There things gets interesting: Asterisk 1.0.RC2 (location 1) didn't had that problem. Then, 2 day ago, I upgraded to 1.0.9, and now I have same problem on BOTH location! Registration to other networks (FWD for example) working with no problems, only registration to each-other is impossible. --- here is configuration: LOCATION 1: [general] register = L1o:[EMAIL PROTECTED] [L2o] type=peer username=L1i auth=rsa outkey=L1 host=dynamic qualify=yes canreinvite=yes disallow=all allow=ilbc trunk=no [L2i] type=user username=L2i auth=rsa inkeys=L2 qualify=yes context=incoming canreinvite=yes disallow=gsm trunk=no LOCATION 2: [general] register = L2o:[EMAIL PROTECTED] [L1o] type=peer host=dynamic username=L2i auth=rsa outkey=L2 qualify=yes canreinvite=yes trunk=no [L1i] username=L1i type=user auth=rsa inkeys=L1 context=incoming qualify=yes canreinvite=yes trunk=no ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Autostart Asterisk on Slackware?
Maybe trivial question, but I cannot find an answer: How to autostart Asterisk (daemon) on Slackware 10? I know that I should put something in /etc/rc.d, but what? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autostart Asterisk (crashing)!
I did, but asterisk won't start when user is not loged in !? rc.local: if [ -x /usr/sbin/asterisk ]; then /usr/sbin/asterisk echo ASTERISK started fi I get echo ASTERISK started when turn on computer, but asterisk is NOT started. When I login as root and type ps -e i get list: PID TTY TIME CMD 1 ?00:00:04 init 2 ?00:00:00 keventd 3 ?00:00:00 ksoftirqd_CPU0 4 ?00:00:00 kswapd 5 ?00:00:00 bdflush 6 ?00:00:00 kupdated 10 ?00:00:00 mdrecoveryd 58 ?00:00:00 syslogd 61 ?00:00:00 klogd 169 ?00:00:00 khubd 521 ?00:00:00 inetd 524 ?00:00:01 sshd 535 ?00:00:00 crond 537 ?00:00:00 atd 540 ?00:00:00 sendmail 543 ?00:00:00 sendmail 553 ?00:00:00 smbd 555 ?00:00:00 nmbd 557 ttyS000:00:00 gpm 564 tty1 00:00:00 agetty 565 tty2 00:00:00 agetty 566 tty3 00:00:00 agetty 567 tty4 00:00:00 agetty 568 tty5 00:00:00 agetty 571 tty6 00:00:00 agetty 576 ?00:00:06 mpg123 579 ?00:00:08 mpg123 583 ?00:00:08 mpg123 587 ?00:00:08 mpg123 622 ?00:00:00 smbd 624 ?00:00:00 sshd 626 pts/000:00:00 bash 639 pts/000:00:00 ps Interesting here is that mpg123 is started from Asterisk, but Asterisk isn't on this list. Seems to me that Asterisk crashed during starting. If I execute /etc/rc.d/rc.local from my root console, Asterisk starting normaly. Why crashing? - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: uto 15. feb 2005 17:01 Subject: Re: [Asterisk-Users] Autostart Asterisk on Slackware? On February 15, 2005 10:49 am, Goran Dj. wrote: How to autostart Asterisk (daemon) on Slackware 10? I know that I sh ould put something in /etc/rc.d, but what? Something like /usr/sbin/asterisk -g in /etc/rc.d/rc.local would do it. You can craft up more complex things if you like, wrap safe_asterisk or do whatver, but that'll get you started. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy's apparantly failing in the field
I am not sure if this is the place for Digium user-to-user discussion, but... We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no dialtone and no registration with the server. Has anyone else experienced high failure rate with these devices? ___ Asterisk-Users mailing list Asterisk-Users@lists.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.conf qualify=yes not working?
Thanks, In these cases, the IAXy cannot be called by another extension. Yes they are behind a firewall router, but in two cases, just a cheapie Linksys WRT54G (or similar Linksys). I also use a WRT54G and do not have this problem with my IAXy. On Wed, 2005-01-12 at 08:15 +0100, Wilson Pickett wrote: In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because iax2 show peers shows the device as status UNKNOWN. However, when a user picks up the telephone plugged into the IAXy, they can place a call just fine within our Asterisk server. Are the IAXy behind firewalls or routers? It sounds like the message sent by asterisk is not geting through. The IAXy would still know how to call out, but can they be called? If UNKNOWN, I assume not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax.conf qualify=yes not working?
We have many IAXy devices in the field now. In all cases, in iax.conf, we have qualify=yes, so that using iax2 show peers, we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because iax2 show peers shows the device as status UNKNOWN. However, when a user picks up the telephone plugged into the IAXy, they can place a call just fine within our Asterisk server. Can anyone tell me if there are any conditions which might affect the functioning of the qualify feature, while still allowing outbound calls to go through? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX2 (IAXy) and DTMF Question
In my particular configuration, by the time the IAXy gets DTMF, it's just audio (e.g. not out-of-band in any way). The SIP modems play the audio of DTMF quite nicely, while the IAXy plays it quite warby, thus my DTMF-driven application (which is plugged into the IAXy) can't decode them. Are there codec settings in the IAXy which might do a better job of rendering DTMF as audio? Thank you, Brent On Tue, 2005-01-04 at 01:56 -0500, [EMAIL PROTECTED] wrote: Looks like the IAXy at the originating end converts the audio DTMF into an IAX DTMF message (and strips the DTMF out in the process). Meanwhile the IAXy at the answering end doesn't convert the DTMF indication message back into tones. FYI, Mark implemented DTMF in the latest version of the IAXy firmware that is in CVS head. It will make it's way to the stable branch at some point. Russell ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 (IAXy) and DTMF Question
I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect (can't hear it, or can only hear clicks). In contrast, when a SIP device calls the Echo test program, we can punch DTMF all day and hear it echoed back to us. Can anyone tell me if we're doing something wrong? Thank you, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX2 and DTMF
For efficiency reliability, when SIP transmits DTMF as non-audio data, it uses RFC2833 or INFO. My question is - (not knowing much about IAX2) - when IAX2 transmits DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it using some other IAX2-specific mechanism with its own name? Thank you, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Command-line dialer/recorder for asterisk?
I'm somewhat new to Asterisk and am tasked with having it perform some automated functions. Is there a way with the current system (and/or extra modules out there to:) 1. Launch something from a command line (on the Asterisk server) to: 2. Dial an extension 3. Issue some DTMF sequences, 4. Record the output to a WAV (or GSM) file, and 5. exit Any quick pointers would be greatly appreciated, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.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-AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear)
My configuration is:
[ISDNPRI] -- [CISCO AccessServer AS5350] --H.323-- [ASTERISK] --
[CISCO ip phone 12SP+/Skinny]
When call is initiated from IP phone - Asterisk - AS5350 - ISDN
everything working ok (RTP is ok).
But, when call coming from ISDN - AS5350 - Asterisk - IP phone
IP phone party can hear ISDN party, but ISDN (incoming) party canNOT
hear IP phone party
because RTP stream from Asterisk is sent to 127.0.0.1 instead to real IP
address of AS5350
Here is H.323 debug, for both situations:
1) ---
--- outgoing call (RTP is ok, both party can hear) --
-- Call token is ip$localhost/12862
-- Call reference is 12862
-- Sending SETUP message
Recieved Open Recieve Channel Ack
=*= In CreateRealTimeLogicalChannel for call 12862
-- externalIpAddress: 10.0.3.15
-- externalPort: 14152
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 1
=-= In OnAlerting for call 12862: sessionId=1
--- found logical channel. Connecting RTP
RTP channel id 1 parameters:
-- remoteIpAddress: 10.10.10.61
-- remotePort: 16862
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 14152
-- Ringing phone for 10.10.10.61
-- Asked to indicate 'Remote end is ringing' condition on channel
Skinny/[EMAIL PROTECTED]
RFC3389: 1 bytes, level 4...
Dec 22 18:51:26 NOTICE[557082]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
=*= In CreateRealTimeLogicalChannel for call 12862
-- externalIpAddress: 10.0.3.15
-- externalPort: 14152
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 2
=-= In OnConnectionEstablished for call 12862
-- Connection Established with 10.10.10.61
-- Asked to indicate 'Stop tone' condition on channel
Skinny/[EMAIL PROTECTED]
=-= In OnReceivedAckPDU for call 12862
channelsOpen = 1
2) ---
---incoming call (RTP misplaced, incoming party don't hear)
Sending alerting
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
=*= In CreateRealTimeLogicalChannel for call 5006
-- externalIpAddress: 10.0.3.15
-- externalPort: 17166
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 1
RTP channel id 1 parameters:
-- remoteIpAddress: 10.10.10.61
-- remotePort: 16700
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 17166
Recieved Open Recieve Channel Ack
answering call
=*= In CreateRealTimeLogicalChannel for call 5006
-- externalIpAddress: 10.0.3.15
-- externalPort: 17166
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 2
RTP channel id 1 parameters:
-- remoteIpAddress: 127.0.0.1
-- remotePort: 2070
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 17166
=-= In OnConnectionEstablished for call 5006
-- Connection Established with 10.10.10.61
-- Received Facility message...
=-= In OnReceivedAckPDU for call 5006
-- Received Facility message...
channelsOpen = 1
___
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[Asterisk-Users] SIP dtmf=rfc2833 not working
We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that the IAXy is digitizing DTMF tones and sending just the pure data, rather than the audio representation, and that this explains why the IAXY's work flawlessly in this application. I am also under the impression that SIP modems should also support a mode like this.. We have tried: dtmfmode=rfc2833 in sip.conf, and we have also tried turning on DTMF Tx: to AVT on the Sipura, but this does not affect reliability at all. So my question is: 1) Are we doing anything wrong, or is there something more we should be doing, to enable DTMF translation (ala rfc2833) in Asterisk and/or our SIP modems? 2) Is there any kind of debugging mode in Asterisk which we can turn on, which will show once and for all whether or not we really have successfully enabled rfc2833? We are using Asterisk 1.0.3, by the way. Thank you very much in advance! Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Optimizing Sipura/Asterisk for DTMF?
We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite well. I was wondering if there are any settings in Asterisk and/or in SIP clients such as the Sipuras, which will optimize the connections for DTMF rather than voice? Thank you in advance, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to tell Who's Online?
On Thu, 2004-12-16 at 17:41 +0100, Wilson Pickett wrote: if I'm missing something obvious, but I couldn't find any console command to show users online. sip show peers iax2 show peers Thank you, Do you know, if an IAXy device (or anything else speaking IAX2) disappears, how long will it be (minutes, hours?) before Asterisk notices they are offline, and iax2 show peers will reflect the change of online status? Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to tell Who's Online?
We have an Asterisk server online, with many SIP clients (some Sipuras, some laptops), and we're also using some IAXy's. I've been trying to find a simple way to check who's online, meaning who is reachable at the moment, without actually going through and dialing everybody. Is there a way to do this with Asterisk? I am sorry if I'm missing something obvious, but I couldn't find any console command to show users online. Thank you, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to process inband DTMF on 2 frames and other messages
Hi, I have Asterisk up and running with one FXO, one FXS and 2 SIP clients (X-Lite). One SIP client is running on laptop (wired connection), and another on PocketPC (5550, WLAN connection). I have dialplan which enables all internal (each user with each user internally) and also external calls via FXO (each client can call outside line). Voice mail, music on hold (MP3) etc all are ok. Still, I have several problems: a) During SIP calls I have the following messages scrolling: dsp.c:1452 ast_dsp_process: Unable to process inband DTMF on 2 frames Does anyone know how to deal with this to eliminate this message? b) When there is no activity in the system, I have another message scrolling (every 15sec or every 1min): chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call some long string or hex numbers@10.10.1.101 for seqno 102 (Request) where 10.10.1.101 is my Asterisk box. Again, does anyone know how to eliminate this message? In addition I have some quality issues with PocketPC calls but I suppose that can not be resolved easily with X-Lite version of the soft client and especially over 802.11b integrated connection. Thanks in advance for any help. Goran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite on PocketPC or WindowsXP
Hi, I just downloaded X-Lite for PocketPC and I am wondering did anyone have it working with Asterisk? I suppose it will be ok, but when I start the application on my PocketPC (iPaq 5550) there is no Manu button to configure the settings for SIP. Can someone help with this issue. Thanks. Also did someone configure it on laptop? Any config files? Goran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Messanger 6.2 with Asterisk
I am trying to configure Messenger 6.2 with * but all the notes how to do it are for older 4.x version of Messenger. Basically, on 6.2 there is no Accounts tab so I cant configure services for account on asterisk. Does anyone have 6.2 working with *. Thanks. Goran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call fails on pulse dial line
I have Digium FXO/FXS card and one of my phone lines is with pulse dialing. At first I didnt have dial tone at all, but after upgrading my Asterisk and Zaptel SW to the latest one (1.0.2) I have dial tone. But, when I try to dial outgoing number it fails after first key pressed. Does anyone know how to solve this? I am in Eastern Europe (Belgrade, Serbia) and our phone lines are mixture of old (pulse) and new ones. So I have 2 lines in my house, one with tone dialing and one with pulse dialing. Is this related to some signaling settings in Zapata.conf? Thanks, Goran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How big .CONF files can be?
I'm new to Asterisk. How big can be sip.conf (and other: iax.conf, extensions.conf...) Is there point when I must use DB (MySQL...) instead of pure .conf? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sccp.so: _use_ast_pthread_create_instead_
I tried to install chan_sccp (make; make install) but after that when asterisk starting: [chan_sccp.so]Sep 25 06:34:28 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: __use_ast_pthread_create_instead__ Sep 25 06:34:28 WARNING[16384]: loader.c:423 load_modules: Loading module chan_sccp.so failed! I tried to replace pthread_create() with ast_pthread_create() in chan_sccp.c, but same error... Help? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
