Re: [asterisk-users] what codec is sip using?
Sean, Try 'sip show channels' or 'sip show channel channelid' for the drill down. I believe the codec in use will be displayed with either command. Dave From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL PROTECTED] Sent: Thursday, September 18, 2008 10:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] what codec is sip using? If you use iax, the console will tell you what codec is being used. But for sip, nothing is shown. With sip debug on, I get: Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722) but I don't see anything that shows which codec was used. How do I find out? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
We've been EXTREMELY happy with the HP 5400ZL series chassis switch. Price per port is about 1/3 that of Cisco when it comes to POE. Price is about $100 per port and all ports are 1Gb with POE by default -- you can't get modules that don't have 1Gb and POE. 10Gb uplinks are available with other modules. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Monday, October 06, 2008 11:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PoE switch recommendations? Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
Obviously we don't need 1Gb connections for VOIP :) Phones support pass through to the desktop and VLAN tagging. The need for 1Gb ports comes from wanting to have 1Gb at the desktop. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Monday, October 06, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? On Mon, 6 Oct 2008, Ken D'Ambrosio wrote: Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. I'm curious as to why you want Gb uplinks on the switches? If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per phone, rather than ~80 - make the sums easier and builds in a margin) 10 calls per Mb/sec. So for a 24-port switch, 24 phones all talking to 24 extensions off that switch, the max the uplink port is going to be pushing out is 2.4Mb/sec. For 200 extensions, say 9 x 24 port switches, with a single top-level (non PoE switch) switch with the PBX plugged in along side the 9 downlinks, that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are in-use at the same time (and the PBX is carrying media) Now you may not want to build the network like that, but it seems that Gb is overkill just for the VoIP side of things. (And with that many extensions, I would suggest keeping all the phones on one set of switches) (Then again, it might not be possible to get big PoE switches without Gb uplinks, so it might be a moot point!) So satisfy my curiosity - why Gb uplinks? Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
Right, it takes some doing to find a 1Gb switching phone though we ended up going with a system based on the Cisco 7941G-GE. This model supports all of the needed features including vlan tagging and 1Gb switching. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Monday, October 06, 2008 12:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] PoE switch recommendations? Most phones support only 100M switching though Unless you run separate cabling for VoIP and data but then you would not need the 1G uplink. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gibbons Sent: Monday, October 06, 2008 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? Obviously we don't need 1Gb connections for VOIP :) Phones support pass through to the desktop and VLAN tagging. The need for 1Gb ports comes from wanting to have 1Gb at the desktop. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Monday, October 06, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? On Mon, 6 Oct 2008, Ken D'Ambrosio wrote: Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. I'm curious as to why you want Gb uplinks on the switches? If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per phone, rather than ~80 - make the sums easier and builds in a margin) 10 calls per Mb/sec. So for a 24-port switch, 24 phones all talking to 24 extensions off that switch, the max the uplink port is going to be pushing out is 2.4Mb/sec. For 200 extensions, say 9 x 24 port switches, with a single top-level (non PoE switch) switch with the PBX plugged in along side the 9 downlinks, that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are in-use at the same time (and the PBX is carrying media) Now you may not want to build the network like that, but it seems that Gb is overkill just for the VoIP side of things. (And with that many extensions, I would suggest keeping all the phones on one set of switches) (Then again, it might not be possible to get big PoE switches without Gb uplinks, so it might be a moot point!) So satisfy my curiosity - why Gb uplinks? Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Did you check sip.conf to make sure that the port is correctly set to 5060? Please show the output of Cli sip show peer peernumber and the contents of your SEPMAC.cnf file. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Wednesday, October 08, 2008 1:20 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized Hi Jerry, Hmm. We had to replace our router with one that supported SIP ALG (we went cisco). However, since you mention all the phones are in the LAN this shouldn't make a difference. Does the problem go away if you go back to the old firewall? Thanks, Matt unfortunately I cannot do that. The other thing I noticed was that doing a sip show peers the port used to show as 5060 and now it has all differnet numbers 49xxx, 60XXX, etc... Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Thank you, Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 3:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan
Re: [asterisk-users] Cisco 7906g SIP
. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
Dare I ask why you want to do this? Dave On Oct 23, 2008, at 10:00 PM, Stephen Reese wrote: I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The second phone I would like to revert back from SIP and connect it to CME and then CME to Asterisk. Is this reasonable or is it a huge pain in the rear? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
Ahh now I see. I am a major proponent of Cisco hardware but it works pretty well with * using either the SIP image or the SCCP image. I would need to have some pretty specific feature needs in order to complicate things with a setup that required CME and * to interact. On the other hand if it's just for fun, that's a different story. And I dare say that it does sound like a fun project to take on. Dave -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Thursday, October 23, 2008 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; David Gibbons Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME) On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote: Dare I ask why you want to do this? Dave I know it seems counter intuitive but I've several examples of it being done and for me it would be for the experience of working with CME. A lot of companies utilize Cisco hardware, I figure why not check it out. I enjoy using Asterisk for my SIP server but there are a number of different configurations out there including using Asterisk as a Voicemail server and Cisco Call Manger as the device to interface with the phone rather then having to flash them and all of that even though I've done it twice and it's not a bad process. Mainly just curious... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decent Voip Phones for enterprise
You can use the Cisco phones with either the SIP of the SCCP image. Though I do agree that the SIP image is a bit easier to setup and auto- provision, the SCCP image is a more native (obviously) implementation. The chan-sccp-b project has nearly every feature usable on these phones working in * at the moment. Dave On Oct 28, 2008, at 8:39 PM, Alex Balashov wrote: You can use Cisco phones as long as they have a SIP image. Kev Szaszvari wrote: Hi there Our company is using the Linksys SPA-942 Phones, and they are pretty useless. They dont have any central management or provisioning, as well as a pretty bad interface. Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have * Central Management for all the phones (We dont mind if we have to buy the software to manage them) * Programable shortcut buttons, So i can program in on certian phones quick dials to queues. * Optional but bonus, The ability to have a shared address book accross the phones. Can i use the Cisco phone managemenet software and the Cisco phones with Asterisk, Or is it 100% cisco. Thanks in advance Regards, Kev The information contained in this e-mail communication may be confidential. You should only read, disclose, re-transmit, copy, distribute, act in reliance on or commercialise the information if you are authorised to do so. If you are not the intended recipient of this e-mail communication, please immediately notify the sender by e-mail and then destroy any electronic or paper copy of this message. Any views expressed in this e-mail communication are those of the individual sender, except where the sender specifically states them to be the views of Mail Call Couriers. Mail Call Couriers does not represent, warrant or guarantee that the integrity of this communication has been maintained nor that the communication is free of errors, virus or interference. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decent Voip Phones for enterprise
Gordon, My guess is that you're a contractor so I can understand why you'd want to keep yourself in high demand by steering clear of the methods that simplify deployment and redeployment. As an employee on the other hand, I want to make things as easy and integrated as I can in order to simplify my own work and keep my employer happy. This mandates the central management features and integration with an existing active directory. Dave --snip-- I always wondered about this - my target is the SME - say 4-150 seats - people don't move desks, change office that often, staff churn is typically low, so I program the phones once then leave them there. If you move desk you take your phone with you. If you leave then the phone can be renamed via it's web interface relatively easily. --snip-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Fair enough, I guess I was concentrating on this line in Jerry's message :) The only reason I can think of not to is to eliminate the cost of the second cable. I believe you're mistaken about the QOS though. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. QOS will absolutely allow voice traffic to pass with priority over heavily loaded links -- this is in fact the reason that it would be implemented. Obviously giving priority to the voice traffic on these heavily loaded links serves to mitigate both latency and jitter. The concern is almost never one of taking bandwidth away from the desktop, but one of the desktop taking bandwidth (especially by introducing latency) away from the phone. Agreed -- but with VLAN tagging and QOS, the issue of how much bandwidth the desktop uses and/or needs becomes moot since the phone is given priority. Dave David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- That's two _logically_ separate networks. The key point is that the last yard cable to the phone is not shared with the computer. The issue is not a lack of bandwidth but that the phone has to try and get its little packets inserted between the massive packets of a database lookup or file transfer in a timely manner (latency and jitter). You might get away with a single logical network on a smaller site or a larger one with very light traffic. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. I only use it on WAN links where bandwidth is more expensive. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 Asterisk
I've never used the Sipura phones but they probably sync with an NTP server. My guess is that the NTP server is on the asterisk box (you can probably verify this by checking the config of the phones and finding the option for NTP server). It is possible that the NTP service isn't running on the asterisk box (after a reboot or a crash) or that the asterisk box's time is incorrect. Do you know what distribution you are running on the server? You can type 'uname -a' at a command prompt and get an idea of the distro. Also try '/etc/init.d/ntpd' start or 'service ntpd start' - these may be able to restart the NTP daemon for you and begin syncing the phones properly again. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Anness Sent: Tuesday, November 04, 2008 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SPA-962 Asterisk Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn't right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that is a bitch when you have many phones, shouldn't the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Steve Anness ICT Support Analyst Humanitarian International Services Group ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
I think I'll take the occasional spam and keep my freedoms and civil liberties... Tell Kim Jong Il I said hello though! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Thursday, November 06, 2008 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. David Gibbons wrote: I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no dial to busy sip line
How about a call queue using the roundrobin strategy?
http://www.voip-info.org/wiki/view/Asterisk+call+queues
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus
Laube
Sent: Friday, November 14, 2008 11:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] no dial to busy sip line
Hi list,
is it possible to get in the running dialplan the status of (SIP) lines
without using AGI or anything like that? What I want is a stepwise
calling: I have several SIP lines (let's say they are three) which I
want to dial to alternatingly. But I do not want to dial to a already
busy line and catch the busy. Instead I do not want to dial to that peer
but to the next one. I want to have a kind of a adaptive dialplan.
Using AGI and such things just makes it slower in my opinion (if I call
an AGI script that does an asterisk -rx 'sip show channels' |gawk -F
{' print $1 '}, for example). Does anyone of you have an idea of how
to do that?
Thanks in advance. Best regards,
Christophorus Laube
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[asterisk-users] Parked Extension Variable
Hello, When I execute parkandannounce() in the dialplan, is the extension that the call is parked to stored in a variable? I would like to send it to an AGI script but can't seem to figure out where the 'announcer' gets its information. Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Extension Variable
snip
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
/snip
*grin*
I guess I deserved that.
Thanks for checking.
Dave
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Re: [asterisk-users] Message 0841984
Last I checked, Lynch mobs don't shoot people. snip I wonder if there would be interest in organizing a bounty for a lynching mob, that would track down these !...@#$# silly excuses for human beings and shoot them. If we all chipped in a few dollars I bet we could hire someone. /snip --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing MP3s...
Ken, An empty conference call or a parking lot with MOHMP3 both come to mind. --Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Thursday, January 08, 2009 4:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Playing MP3s... For no reason other than it would be cool, I'd like to be able to dial an extension and have it play a random MP3. Since, however, MP3s are kinda-sorta weird due to patents, I'm not sure what the right approach for this is. Any pointers on how to go about this? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Stealing
I'm confused as to why you think leaving a phone off the hook is better than parking the call and hanging up the phone. The phone that's off the hook can't receive any more calls after you've 'pulled' the one it was on the line with, assuming you don't walk back to that phone and subsequently hang it up, making the originating extension effectively useless. Call parking and hanging up the originating extension is actually a more elegant solution in my opinion. --Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane Sent: Thursday, January 15, 2009 3:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Stealing You just leave the phone off the hook, walk to the handset to which you want to transfer the call, then dial the call-steal code. This steals (captures) any active call within the same ring group. You don't need to park the call first. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting observation
snip My understanding is that Charter 'telephone' doesn't use IP at all but rather uses some additional frequency spectrum on their cable network. Hence, the reason why faxing with their service is reliable unlike other providers who are *actually* using VoIP. /snip I think what you're referring to is the general hesitance of the cable providers to call their phone service VOIP service. VOIP still has a negative connotation with most regular folks, so they don't want to negative PR. I'm don't have any facts, but I'll bet you a penny that they don't have a proprietary system using something /OTHER/ than IP to send encapsulated voice over 'additional frequency spectrum'. That would be prohibitively expensive to develop and pointless from a technical standpoint, given that IP telephony is already set to deploy and relatively mature. The reliability of faxing is based soley on network jitter and latency and codec compression. I've found that taking the compression out of the mix (using g.711 ulaw) and controlling the jitter and latency (something that's easy to do on a private network like theirs with QOS) causes faxing to be pretty darn reliable. --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting observation
snip I'd be willing to bet *TWO* pennies that you're correct. I certainly was not coming into the conversation as an expert, just stating what I'd read/heard of their service... hence the My understanding is that... beginning to the email. :-) /snip Fair enough. I get worked up when I hear the cable companies calling their phone service anything other than VOIP :). I'm going to hold off on going on a 2-page rant about the cable companies, their false advertising, awful performance, sub-par quality and terrible customer service. Then again, I heard Verizon has been known to burn down your house when they install FiOS... Yikes! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for Asterisk experts
snip One problem to overcome is that your competitors are: 1) Literate. 2) Post to the right mailing lists. Meftah Tayeb wrote: /snip Ha ha ha ha. So, you're saying you don't want the job? LOL. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
The higher you raise the barrier for entry to the mailing list, the more you decrease the amount good the mailing list is actually capable of doing. (barrier height is inversely related to how much help we can provide to the people that need help the most) I agree with you regarding the subject spelling/misspelling as it pertains to indexing on the search engines, etc. But if you require those posting to jump through your *10* hoops for the first *10* times they post something (yes, that's 100 hoops. I'm tired of jumping already), you are artificially limiting the number of users that this list can actually help. I don't like getting broken English replies and questions that don't make any sense any more than the next person, but I also get a good chuckle out of reading them. And reading replies that tell people to 'rm -rf /*' gives me a good laugh, too. The only way to REALLY learn is to make mistakes, even if you're making those mistakes because you took the 'advice' that someone gave you for free on the mailing list... Give me a break :) Mailing lists are supposed to be fun and get off topic sometimes. That's what makes them interesting. --Dave PS: Can anyone help me with my broken *.? the ntework card is blinking red and the sips are dropping with echoes. Tai? LOL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, January 27, 2009 10:58 AM To: Asterisk Users Mailing List Subject: [asterisk-users] RFC -- Improving the quality of the mailing lists The -user and -dev mailing lists are a valuable resource -- when they are not cluttered by posts unrelated to the charter of the lists. In my limited memory, this last weekend represents a new low in the relevant subject to noise ratio. Replying to requests with meaningless, misleading, or misspelled subject lines (I need help, asterisk help, Ntework Card) encourage careless posting and obfuscate useful replies from search engines. Also, while replying to such requests may seem helpful, some of the requests indicate such a lack of basic understanding that giving the answer is like giving a small child a very sharp knife when they ask for a slice of bread. For example: How do I delete these files that end in that squiggly thing in my current directory and all directories below? Since most of these users are probably running as root, a simple extra space here and a missed character there (rm --force --recursive /* ~ vs rm --force --recursive ./*~ can have catastrophic consequences. In an attempt to improve the quality of the lists, I propose the following: For a user's first 10 posts, they will receive a reply with a link to a web page and have to answer the following questions: 0) I acknowledge that I am asking for free help and I acknowledge that following the conventions below increase my chances of engaging another list member with relevant expertise and resolving my request. 1) I am posting a new request. a) My request cannot be answered on a more general list such as Beginning Unix, or on a distribution specific list. b) My request cannot be answered on a more specific list such as an AsteriskNow or Trixbox list. c) I have attempted to search for an answer using a search engine such as Google. d) I know what thread hijacking is and I created this request from scratch. e) I have created a meaningful subject line that indicates with as much specificity as reasonable which part of Asterisk I need help with and why. f) I am not posting a self-serving message directing someone to my product that would be better posted to the -biz list. g) I am not posting in HTML. h) I am posting in English. i) I am fluent in English or I have attempted to have someone who is review my request. j) I have run my request through my spell checking resources. or 2) I am posting a reply to a post. a) I know what top posting is and I am not ignoring the convention of the list. b) I am not posting a self-serving message directing someone to my product that would be better posted to the -biz list or only to the requester. c) I am not posting in HTML. d) I am posting in English. e) I am fluent in English or I have attempted to have someone who is review my post. f) I have trimmed the previous post down to just the point(s) I am replying to. g) I have run my request through my spell checking resources. For -dev, the following questions would be added: ) My post directly relates to changes in the Asterisk C source code. ) I am not reporting a bug or a posting a patch that should be directed to bugs.digium.com. Included in the web page would be the original message with the ability to change the list the message is to be posted to, the subject line, and the body of the message. Comments? Thanks in advance, Steve Edwards sedwa...@sedwards.com
Re: [asterisk-users] RFC -- Improving the quality of the mailing lists
How pompous are we now? What happened to the 'open source community'? There's a give and take involved; you answer questions you know how to answer in the hopes that someone with greater experience and knowledge of the software will answer your questions. Yikes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Tuesday, January 27, 2009 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RFC -- Improving the quality of the mailing lists At 09:30 AM 1/27/2009, you wrote: People are always going to ask stupid questions. For me it's not so much the stupid questions as the expectations that we're here to solve their problems according to their needs. If that continues to happen and the noise level gets high enough those that have the most to offer will leave and all will be lost. Maybe there needs to be a beginner list and posting on this becomes invite only from people who participate on that list. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
If your provider has two different IP addresses at its endpoint, you could use iproute2 (source based routing) with two local source addresses to make sure that there is a one-to-one mapping of source address to destination address. Then you could have two peer definitions and an address=declaration in each. As I think about it, I believe that with iproute2, you could use one provider endpoint address and two local addresses in the same manner, without the one-to-one mapping... This seems like the most elegant solution in my mind. And the only one that will work reliably... :) --Dave snip If that code in the below link worked, will I be able to have two SIP (IP Trunk), both send for same destination IP:Port, but from different source IP's? So the destination will authorize me in my two different IP's? /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Reply to Our Critical Packet SIP Calls Dropped in Voicemail
Which firmware load? We had all kinds of trouble with 8.4.x, after being stable for a few months on 8.3.x. Going back to 8.3.x made all of the weirdness disappear. While we're on the cisco note, I have script to remotely reboot the SIP firmware load Ciscos and to provision the phones based on active directory if you're interested... back on topic: Have you run a packet cap on a mirror of the switchport the phone this is happening on is connected to? Anything strange? What's happening on the switch backplane (network backbone) at large when you notice the problems? Major transfers/lots of traffic? Anything else running on the * server? --Dave snip We're running Asterisk 1.4.22 built from source and Cisco 7961G phones with the SIP firmware image. I've tried most of the recent firmware versions for the phones with no real impact on the issue. Strange thing is that while all of the phones use a variation on the same config file (with the only changes being the SIP account details and speed dial keys) but one user in particular seems to suffer the issue far more frequently. /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Have you tried configuring two peer config files and setting the externip parameter in each of them differently to your two public ips? Dave -Original Message- From: bilal ghayyad [mailto:bilmar...@yahoo.com] Sent: Monday, February 02, 2009 2:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; David Gibbons Subject: RE: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine My provider has one IP and one port ONLY, I need to send for him the calls from different IP's on the same Asterisk machine, how? Regards Bilal --- On Mon, 2/2/09, David Gibbons d...@videon-central.com wrote: From: David Gibbons d...@videon-central.com Subject: RE: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: 'bilmar...@yahoo.com' bilmar...@yahoo.com, 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Monday, February 2, 2009, 2:16 PM If your provider has two different IP addresses at its endpoint, you could use iproute2 (source based routing) with two local source addresses to make sure that there is a one-to-one mapping of source address to destination address. Then you could have two peer definitions and an address=declaration in each. As I think about it, I believe that with iproute2, you could use one provider endpoint address and two local addresses in the same manner, without the one-to-one mapping... This seems like the most elegant solution in my mind. And the only one that will work reliably... :) --Dave snip If that code in the below link worked, will I be able to have two SIP (IP Trunk), both send for same destination IP:Port, but from different source IP's? So the destination will authorize me in my two different IP's? /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
snip Problem is that its crashing for seemingly no reason at all, no errors on the console, no logs (that I can find), nothing in /var/lib/messages - its puzzeling! Management is screaming like banshees, calls are dropping like flies, and all hell is about to break loose if I can't stop asterisk from crashing every couple of hours, taking down any Zaptel calls with it. /snip I am assuming you have debug turned on so that you can see what's going on when it crashes? If not, open the * console (asterisk -r) and type (core set verbose 100) and (core set debug 100). Then leave the console open so you can see if * was doing anything special when it crashed. --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?
snip Anyone use CIsco 1760 with Asterisk /snip No, but I'm using 7941G-GE and 7961G-GE in a deployment of ~80 phones. Did you have a question about implementation or are you just curious? --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phone 7940G.
I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment. I'll provide SEPMAC.cnf.xml's if requested off-list. --Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, February 13, 2009 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phone 7940G. Catalin S. wrote: hey finally i did it. I upgraded the firmware to the latest sip firmware and now i have the another problem. The requested files are the following: ---///--- Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf to 192.168.1.3:51253 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to 192.168.1.3:51254 ---///--- I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is the mac address of phone, but i don't know what to write in CTLSEP00141CAA4B4C.tlv, Create an empty file and it will be happy. At least that has been my experience with my 7960. Others can probably provide a sample of the remaining files. So far I have been unable to go beyond version 7 firmware, as it is unhappy with the XML file when trying to move to version 8. John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phone 7940G.
snip On a similar subject, I have been able to get a 7961 to switch to a SIP firmware, has anyone had any luck with this? /snip Yes, I have several 7961s and 7971s running SIP, same firmware generation as the 41s --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
snip Certainly a sobering thought. Have others had to deal with this in PBX replacement scenarios? Its a giant cost savings in this case - they are dropping about 12 POTS lines in favor of utilizing (an underutilized) T1 trunk that was already in place. /snip Yes -- our alarm monitoring company considers T1 - * - ATA - Alarm to be so unreliable that they require you to sign a waiver (indemnifying them in the event of basically anything) if you hook it up this way. Because of that we kept a POTS line around to hook up the alarm system. It would be cheaper to hook the alarm panel up to an internal cell phone backup :). I assume there are manufacturers that offer a built-in cell modem... --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
snip We will be testing the ADT connection heavily this week. The modem connections to my understanding are 2400 baud. Over G.711U and a T1 I don't see why this wouldn't be as solid as a POTS line, but our tests will tell! /snip We do *fax* in this way and it works like a charm. We can hit much more than 2400 baud I think too. --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing SIP hardphones call history
You could use the XML browser on the cisco 79xx series. --Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Chamberlain Sent: Thursday, February 19, 2009 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Managing SIP hardphones call history On Feb 18, 2009, at 10:47 PM, Olivier wrote: Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the hardphone which query to send and then, you would get a customized Call History. The Cisco SPA962 and SPA525 support RSS feeds, you could do a call history RSS feed for each phone. -- Eric Chamberlain, Founder RF.com - http://RF.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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call from '6000' to extension rejected because extension not found
Is this a question? Haha. Computer won't doesn't turn on. Got blck scrn. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman Sent: Wednesday, February 25, 2009 3:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call from '6000' to extension rejected because extension not found Call from '6000' to extension 'xx' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk with 250 lines
I have several Dell boxes running onboard Broadcom and Intel NICs any haven't had any issues. It's preposterous to make a blanket statement like that about all Dell hardware. Maybe you should re-compile your drivers. Or have prosupport come put a new mobo in for you :). -Dave snip Not at all, just Dell :) /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI dropping #2
Harry, Chill on the duplicate posts. Sometimes the listserv takes time to forward the message. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harry Vangberg Sent: Thursday, March 26, 2009 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI dropping #2 Okay. Trying third time, sorry for that, might just be my mail client, anyways, from voip-info.org: RED: Loss of signal (LOS): The equipment shall assume loss of signal when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS. This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their amplitude, theirs being 2048. Does this make *any* sense? 2009/3/26 Harry Vangberg ha...@vangberg.name: Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf loadzone=dk defaultzone=dk span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 == == /etc/asterisk/zapata.conf [channels] switchtype=euroisdn usecallerid=yes group=1 signalling=pri_cpe context=incoming channel=1-15 channel=17-31 == The Asterisk console has this (repeating for every channel): [Mar 26 18:39:19] WARNING[3772]: chan_zap.c:6685 handle_init_event: Detected alarm on channel 1: Red Alarm [Mar 26 18:39:19] WARNING[3772]: chan_zap.c:1471 zt_disable_ec: Unable to disable echo cancellation on channel 1 [Mar 26 18:39:19] WARNING[3772]: chan_zap.c:6685 handle_init_event: Detected alarm on channel 2: Red Alarm [Mar 26 18:39:19] WARNING[3772]: chan_zap.c:1471 zt_disable_ec: Unable to disable echo cancellation on channel 2 ... ... [Mar 26 18:39:19] NOTICE[3771]: chan_zap.c:8486 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 [Mar 26 18:39:19] WARNING[3771]: chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Mar 26 18:39:24] NOTICE[3771]: chan_zap.c:8486 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 [Mar 26 18:39:24] NOTICE[3772]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 1 [Mar 26 18:39:24] NOTICE[3772]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 2 ... ... See the full output at http://sprunge.us/cdFf I enabled PRI debugging for span 1, which gives this: q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH -- Got UA from network peer Link up. q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED q931.c:2755 q931_restart: call 32768 on channel 1 enters state 62 (Restart) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Terminator) Message type: RESTART ACKNOWLEDGE (78) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 121 (cs0, Restart Indicator) q931.c:3581 q931_receive: call 32768 on channel 1 enters state 0 (Null) q931.c:2755 q931_restart: call 32768 on channel 2 enters state 62 (Restart) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2
Re: [asterisk-users] ATT PRI Install - What is outpulsed?
This is the outgoing callerid. If you have 1200 DIDs in a range, you probably only need to outpulse 4 digits (they already know the first six). If you want to be able to make your callerid anything that may or may not be one of your DIDs, you probably want 7 or 10. I pick 10 no matter what for the flexibility. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Friday, March 27, 2009 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ATT PRI Install - What is outpulsed? Hey All, ATT is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have any idea what she was referring to? Is this ANI? Outgoing Caller ID? Something else? While I've done many POTS line setups, this is my first PRI install, so I'd also welcome any make sure you do this, read this first or ATT always messes this up so... tips. Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no ringtone - just silence/bridging of external calls
I had a similar situation a while ago and the fix was setting up indications.conf: http://www.voip-info.org/wiki-Asterisk+config+indications.conf -Dave snip I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same problem than me and a resolution for it? /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice SIP 183 progessl
The fact that you sent this again (what is that -- 3 times now?) AND with high
importance, will likely cause people to ignore your messages rather than trying
to help you.
There are few things that annoy me more than messages sent with high importance
(same category of annoyance as messages written in all caps). Let's have a
little bit of intarweb etiquette.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab
Sent: Friday, April 03, 2009 9:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Importance: High
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
or if there any way to stop the music on hold and let the caller hear the
Ring Back Tone
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
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Xplorium does not guarantee the integrity of this electronic message and any
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defects.
*
___
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asterisk-users mailing list
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No employee or agent
Re: [asterisk-users] Please Advice SIP 183 progessl
Lol.
I'm actually in the small minority who prefers top posting to bottom posting.
-d
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, April 03, 2009 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
And don't top post ;)
On 3 Apr 2009, at 14:38, David Gibbons wrote:
The fact that you sent this again (what is that -- 3 times now?) AND
with high importance, will likely cause people to ignore your
messages rather than trying to help you.
There are few things that annoy me more than messages sent with high
importance (same category of annoyance as messages written in all
caps). Let's have a little bit of intarweb etiquette.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
] On Behalf Of Khaled W. Chehab
Sent: Friday, April 03, 2009 9:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Importance: High
Dears
Kindly find my dial script below,I am trying to send the caller 180
ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
or if there any way to stop the music on hold and let the caller
hear the
Ring Back Tone
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no
answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my
source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement
on
behalf of Xplorium with another party by e-mail without express
written
confirmation by an officer of Xplorium. Any views expressed by an
individual
in this electronic message do not necessarily reflect views of
Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to
the
addressee(s), and contain confidential information protected from
disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and
its
attachments, kindly delete it immediately from your system and
notify the
sender by electronic mail. You must not copy this message or
attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message
and any
of its attachments, or that they are free from computer viruses or
other
defects
Re: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?
Yes, you can flash them back and forth as you require. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Shauger Sent: Monday, May 04, 2009 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP? Anyone know if we take a Cisco phone off of a Call Manager system and flash it for SIP to demo on Asterisk, can we take it back to Cisco and Call Manager will remember its MAC address and reflash it back to what it is supposed to be? I would anticipate with Cisco Discovery Protocol this would be the case, but would like to be sure. Thanks! David Shauger Vice President Sollos Technology Solutions 678-317-9444 - voice 404-886-7603 - cell 772-679-5830 - fax d...@sollos.commailto:d...@sollos.com http://www.sollos.com/ This email has been certified by Thawte Email certification helps prevent identity theft Virus scanning provided by Clam Antivirus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
snip ...routing via satellite adds about a quarter second of latency to the path. Is that too much? /snip Eric, I believe that you are mistaken. Routing via satellite adds about a quarter second of latency PER TRIP from earth to orbit. This is simply due to the distance a satellite is from the ground and the speed of light (interference not withstanding). Traceroutes and pings to satellite providers can be misleading because they cache some content on the birds in order to decrease latency. As I recall they even intercept some pings to accomplish the same. A *real* round trip for a VOIP call and/or non-interfered TCP connection would look like this: 1. Your device up to the bird (~250ms) 2. The bird back to the ground (~250ms) 3. The ground station out to the internet (~Nms) 4. The internet back to the ground station (~Nms) 5. The ground station back to the bird (~250ms) 6. The bird back to your device (~250ms) As you can see, even the one way udp stream will take approximately 500ms beyond any latency introduced by things such as your wireless network and the internet. VOIP over satellite, as Josh indicated, will be painful. You'll be talking all over one another due to the delay assuming that the stream can even be sustained with that much latency. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
snipOf course, that's assuming your satellite is in geosynchronous orbit. If It's in LEO.../snip Singer, You are of course correct, low earth orbit will have lower latency. I was assuming that this user would be using a stationary link on the ground, not a portable sat phone or an aimable dish to make these calls. That may be an incorrect assumption. Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proxying from one server to another
Redirect traffic with iptables like this: Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to NEW_PUBLIC_IP I'm not sure if this will work for SIP. You may need the proxy to change info in the sip messages between server and client. --Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Wednesday, May 13, 2009 8:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Proxying from one server to another Hi All, I'm trying to find a software package to do the following sip proxy work: I've an A*k server A that needs to be decommissioned, from the USA, and replaced by server B, in the UK. Both servers are on public internet IPs. Whilst the client migration happens, I want to divert all the Register traffic from Server A to Server B to catch any clients still left out there. Unfortunately, the original Clients were configured with static IPs instead of DNS names for the SIP Registrar, so I have to proxy Server A until all the clients have been updated (which might be a long time). Obviously A*k itself wont do this (as far as I know). I've looked at siproxyd and party-sip, but with no success so far. I've also tried using IPtables to redirect at the IP level, but the public IP ranges seem to stop me from achieving this. It works in my local-lan testing, but not on the public servers. Any ideas? Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail and remote directory with SSHFS
Tunnel samba or nfs through ssh, rather than using sshfs, then mount using once of those more ubiquitous standards. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Wednesday, May 13, 2009 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail and remote directory with SSHFS Hello! I am trying to mount a remote directory for voicemail using sshfs. However, whenever Asterisk attempts to write the file, it fails, because SSHFS cannot lock the directory. Is there a solution to this problem or an alternative method for using a remote directory for voicemail? Thanks, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting Cisco 7961 to SIP
Ahh I see. In response to your other question about the auto-provisioning of Cisco phones, I wrote some scripts that work against an active directory and setup the phones automagically. I'll send the link your way if you'd like. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Did not mean to infer they don't perform wonderfully with Asterisk. By hack I meant that Cisco does not offer any official support for them on Asterisk. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will essentially be a hack. I am not sure how proprietary the Avaya system is in regards to registration and open-SIP support. Asterisk and any iteration of it will support it, but Cisco hasn't really designed a load compatible with it yet. I can tell you that I haven't really found any configuration file generation tools for these files. The reason being is that these loads are mainly used for SCCP and SIP Cisco systems. There is a well known tutorial on how to Hack to the CP-7970 to trixbox CE located here: http://www.asterisktutorials.com/cisco-7970-ip-phone/ This may help get you jump started and pointed in the right direction. The only problem that may arise is that in the tutorial, the use a specific SIP load (8.0.3) which may not be available for the 7961G. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
I could be wrong but I don't think the cat5 limit of 100 meters applies to any analog signaling over that copper. I believe it only applies to Ethernet signaling. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, May 26, 2009 10:41 AM To: bald...@rogg.is; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS port The best a native cat5 can run is 100 meters. Unless you like paying your telco huge bucks, you should go for some kind of SIP connection to your box. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-us...@rogg.is Sent: Tuesday, May 26, 2009 9:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum cable length for analog phone from FXS port Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting Cisco 7961 to SIP
Cory, Precisely what do you mean by 'Anything other than Callmanager will essentially be a hack'? I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP image. They're not 'hacked', they're set up properly against the Cisco provided SIP image and are rock-solid stable. I would pit them against any of the cheaper model SIP phones any time, any place, any day. I've written scripts to do nearly everything that call manager can do without paying hundreds of dollars per user for the call manager software. Just about the only thing they can't do at the moment is BLF because they require SIP over TCP to handle SIP messages about BLF status, something that I'm not willing to implement just yet. In the past, Cisco phones have had a bad rap as not being usable outside of a call manager environment. That's just not the case. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Darrin, The files you are using are consistent with SIP for Cisco Call Manager. Anything other than Callmanager will essentially be a hack. I am not sure how proprietary the Avaya system is in regards to registration and open-SIP support. Asterisk and any iteration of it will support it, but Cisco hasn't really designed a load compatible with it yet. I can tell you that I haven't really found any configuration file generation tools for these files. The reason being is that these loads are mainly used for SCCP and SIP Cisco systems. There is a well known tutorial on how to Hack to the CP-7970 to trixbox CE located here: http://www.asterisktutorials.com/cisco-7970-ip-phone/ This may help get you jump started and pointed in the right direction. The only problem that may arise is that in the tutorial, the use a specific SIP load (8.0.3) which may not be available for the 7961G. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw Sent: Tuesday, May 26, 2009 7:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Converting Cisco 7961 to SIP As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn SIP41.8-4-4S.loads term41.default.loads term61.default.loads Now I've read over loads of documentation on it, but am getting tripped up. Most of what I've seen talks about the older firmware versions usually 7.4. I have a feeling I'm still missing a lot of stuff. Anyone have any recent links or information? Also, anyone know of a decent way to generate the config files? I'd hate to have to go through all of it manually? Thanks. Cheers, Darrin Henshaw This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ --
Re: [asterisk-users] SIP over VPN
Assuming you mean the firewall in front of the client, you don't need to open any ports as long as the VPN client is tunneling all traffic to and from the Asterisk server. I set NAT=yes in the config file for the extensions behind a VPN. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo Sent: Tuesday, May 26, 2009 11:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP over VPN Hi all, I have a question. I have a VPN and I want to use a SIP softphone on my notebook using with asterisk. But I have some problem with firewall and port. Someone knows which ports I should open on my firewall??? I can't connect ... Thanks all. Marco ___ -- 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] Cisco 79xx Scripts [WAS: Converting Cisco 7961 to SIP]
Alright, by popular demand, here they are: All of the scripts are written in PHP (so I'm kind of partial :) and you'll need to have it compiled on the asterisk box with cli and ldap to hook into AD and be useable from the command line. Beyond the auto-provisioning script, the remote reboot script is the one that's really a big deal. As far as I know, this is the first method anyone has come up with to reboot a Cisco phone without call manager gracefully and remotely. Cool! Here is the auto-provisioning script: http://www.dave.vc/wordpress/?p=38 Direct Download: http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip Here is remote-reboot script: http://www.dave.vc/wordpress/?p=14 Direct Download: http://dave.vc/wordpress/wp-content/uploads/2008/09/phonepush.zip I also have some weather scripts for the phone browser and AD directory for the phone browser. I'll put them up sometime here... -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Thursday, May 28, 2009 5:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP I'd like to see that link too! I use Cisco 7940s at the moment, and would like to see how to hook them into AD -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: 26 May 2009 15:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Please do! Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, May 26, 2009 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Ahh I see. In response to your other question about the auto-provisioning of Cisco phones, I wrote some scripts that work against an active directory and setup the phones automagically. I'll send the link your way if you'd like. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews Sent: Tuesday, May 26, 2009 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP Did not mean to infer they don't perform wonderfully with Asterisk. By hack I meant that Cisco does not offer any official support for them on Asterisk. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil
Re: [asterisk-users] Suddenly the voice became garbage(likerobot)using Asterisk 1.4.19.2
Danny, Just out of curiosity, can you elaborate? Anything in use for asterisk should be in cache by the time it's needed for a SIP stream. And nothing related to a SIP stream should ever be read directly from the disk... Unless I'm mistaken. Thanks Dave snip Since this is internal SIP, I'd probably vote for a memory leak, bandwidth problem or hardware hiccup. I've had a similar situation when a grep caused pounding of a bad disk sector. /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
I think you're overlooking your internet uplink, which is what I'm talking about: snip Inbound calls are via in IAX trunk from Gradwell. /snip You certainly DO need QOS to maintain call quality over the INTERNET link. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Call quality - how to debug
Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? snip So theres no web browsing etc on that 2mb circuit. /snip In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941G Auth
I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941G Auth
Hey Sasa, I have templates of all the files you need here (SEP file, extension file): http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip If you need further assistance, let me know. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sasa Sent: Monday, June 22, 2009 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Jonathan Thurman wrote: What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device Hi, I have already writed also in SEPMacAddress.cnf.xml file (other at XMLDefault.cnf.xml file) the parameter: loadInformationSIP41.8-0-2SR1S/loadInformation ..but the problem isn't resolved !. Can I try to change some parameters ?..are desperate ! I think I have tried everything ! Thanks. -- Salvatore. - Original Message - From: Jonathan Thurman jthurma...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 6:04 PM Subject: Re: [asterisk-users] Cisco 7941G Auth What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device And you shouldn't need the tlv file. -Jonathan On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843
Re: [asterisk-users] Removing line 2 from CISCO 7940g
Mike, 1. Remove the 'line 2' entries completely from the SEPXX.XML file. 2. Change the 'Version' tag in the SEPXX.XML file. You need only change one digit; I usually just increment the last digit. (version1.0.0.0-9/version). 3. Restart the phone (Settings - **#**). 4. This should do it. If it doesn't, proceed to step 5 with caution. 5. If the line still appears, reset the phone to factory defaults (Hold # while booting, then dial 123456789*0# when the line lights flash amber back and forth). DO NOT RESET TO FACTORY DEFAULTS IF YOU DON'T HAVE THE TFTP SERVER SETUP WITH THE FIRMWARE IMAGES. This will force the phone to re-download the SEP.XML file. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 24, 2009 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Removing line 2 from CISCO 7940g Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message Waiting Indicator on DAHDI line
This may be a stupid question, but IS THERE a message waiting against your PSTN
lines?
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, August 04, 2009 1:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Message Waiting Indicator on DAHDI line
Folks,
I have recently installed Asterisk 1.6.1.1. I have two PSTN lines connected to
a TDM400 and two VoIP lines using SIP. I have a CISCO 7940 using SIP as my
desk phone. Calling any of the four lines should ring the desk phone. This
works fine, except that when ringing the PSTN lines, it activates the MWI on
the 7940. I can see this happening on the console:
[Aug 4 16:48:47] NOTICE[2964]: chan_dahdi.c:7669 ss_thread: MWI:
Channel 3 message waiting!
Looking at the offending piece of code, it seems to suggest from the comment
that it is getting the MWI from the CLID.
/* If the CID had Message waiting payload, assume that this for MWI only and
hangup the call */ if (flags CID_MSGWAITING) {
ast_log(LOG_NOTICE, MWI: Channel %d message waiting!\n, p-channel);
notify_message(p-mailbox, 1);
/* If generated using Ring Pulse Alert, then ring has been answered as
a call and needs to be hungup */
if (p-mwimonitor_rpas) {
ast_hangup(chan);
return NULL;
}
}
I have set usecallerid=no on both interfaces and globally but I still cannot
get it to stop.
I have failed to turn anything up on Google regarding this.
Does anyone have any suggestions please?
Mike.
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Re: [asterisk-users] Asterisk don't detects hang-up by phone
I was having the same problem with about half of my POTS lines. I switched the polarity on the connections for those lines and the problem disappeared. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Thursday, August 06, 2009 9:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk don't detects hang-up by phone Assuming you are connected to a regular phone line, the hang up signal from the phone line would be a break or reversal of polarity of the DC signal on the phone line. (We connect to PRIs, so our signaling is on a data channel. I assume you don't. ) The first question you need to answer is Are you getting a voltage drop or polarity reversal when the other end disconnects? Asterisk has to have a signal to respond to. Some Telcos may not give that signal. Check your phone line with a meter. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL Sent: Thursday, August 06, 2009 7:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk dont detects hangup by phone Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring Asterisk uptime
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need host keys to do
this without user interation), you should be able to poll any linux server for
really anything you want.
Someone who is in the business of selling hosted applications should be able to
EASILY use awk and grep to figure out if his sip trunks are in the 'Registered'
state via SSH... Unless I misunderstood the nature of your question and you
were looking for something native to asterisk or the AMI.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham
Sent: Thursday, August 06, 2009 10:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitoring Asterisk uptime
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server availability.
I'd like to add Asterisk to the servers that our monitoring systems are
handling, and also that our SIP trunk provider has our Asterisk system
correctly registered at all times.
What are the 'best practice' tricks used for monitoring an Asterisk
phone system for uptime and SIP registration from an external monitoring
server? I can certainly ping the box, but I really need more than this.
I need to know if the Asterisk service is running, and also that there
hasn't been any issues with SIP registration to our external trunks.
If anyone could share how they are doing this sort of thing, it would be
greatly appreciated.
Thanks
Myles
--
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
http://www.techsolusa.com
Phone +1-480-451-7440
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Re: [asterisk-users] Cisco 1760 Multiline phone
Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can't register to one 'extension' afaik) and ring them in order: 1,1,Dial(SIP/xx) 1,n,Dial(SIP/xx1) 1,n,Dial(SIP/xx2) Or ring them at the same time: 1,1,Dial(SIP/xxSIP/xx1SIP/xx2) Someone else may have better solution though. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Sorry I mean to say cisco 7960 phone. From: Jimmy Ezell Sent: Tuesday, August 11, 2009 9:15 AM To: 'asterisk-users@lists.digium.com' Subject: Cisco 1760 Multiline phone I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file = [incomming1] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming2] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming3] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 === Jimmy Ezell Assistant IT Manager (408) 487-2200 [cid:image001.jpg@01CA1A84.58CF8240]http://www.hmhca.com/ inline: image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Multiline phone
Jimmy, To clarify, you want to configure the phones like this where p means phone and l means logical line: Phone 1: P1l1 P1l2 P1l3 Phone 2: P2l1 P2l2 P2l3 Phone 3: P3l1 P3l2 P3l3 It sounds like (and looks like) you're dialing all of the extensions on one phone at the same time, which is why they're ringing and ringing. What you want to do is place the extensions for line 1 of each phone (p1l1,p2l1,p3l1) in the dial command to ring them simultaneously. asterisk will then fail through if none of the phones answer in time. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Multiline phone Thanks for the help, I really appreciate the feedback. I tried ringing them all at the same time as you suggested: exten = workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomming4SIP/incomming5) but it does very strange stuff: - I have to push the extension button twice to answer. - More then one extension shows off hook at the same time (Maybe 2 or 3 of the 5 will show off hook on the phone) - When I hang up the phone starts to ring again even though there is no caller I tried ringing them in order: exten = workhours,1,Dial(SIP/incomming1,5,r) exten = workhours,n,Dial(SIP/incomming2,5,r) exten = workhours,n,Dial(SIP/incomming3,5,r) exten = workhours,n,Dial(SIP/incomming4,5,r) exten = workhours,n,Dial(SIP/incomming5,5,r) exten = workhours,n,Macro(voicemail,100) Now I see the call march along each of the extensions until it gets to the end goes to voice mail. What I really want is for the call to go to only one of the unused lines and then fall straight through to voicemail after the timeout. Anyone have some thoughts on getting it to work that way? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, August 11, 2009 10:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can't register to one 'extension' afaik) and ring them in order: 1,1,Dial(SIP/xx) 1,n,Dial(SIP/xx1) 1,n,Dial(SIP/xx2) Or ring them at the same time: 1,1,Dial(SIP/xxSIP/xx1SIP/xx2) Someone else may have better solution though. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Sorry I mean to say cisco 7960 phone. From: Jimmy Ezell Sent: Tuesday, August 11, 2009 9:15 AM To: 'asterisk-users@lists.digium.com' Subject: Cisco 1760 Multiline phone I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file = [incomming1] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming2] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming3] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 === Jimmy Ezell Assistant IT Manager (408) 487-2200 [cid:image001.jpg@01CA1A99.E2624550]http://www.hmhca.com/ inline: image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79XX, SIP and Asterisk
I am using the phones quite successfully, though I have not tried non-English menus. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, August 12, 2009 12:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 79XX, SIP and Asterisk Hi, Is anyone successfully using SIP-enabled Cisco 79XX phones with Asterisk ? Could you then configure this phone to display non-english menus (in french, spanish, german, ...) ? Mine is using a rather old SIP firmware (8.3 ?) with which I could get non-english menus. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Twitter is Suing me!!!
I fail to see how Obama has ANYTHING to do with this. Danny, please DO elaborate so that I don't have to go on believing that you're a fool. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, August 12, 2009 1:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Twitter is Suing me!!! Don't count yourself as out of the woods yet... They will at best make your product inoperable by denying your existing client base access, then may still come back at you. I'd be prepared to launch a second and subsequent product that does the same thing if needed. Welcome to Obamerica! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: Wednesday, August 12, 2009 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Twitter is Suing me!!! Yeh I'm starting to learn the difference - sorry first time I've ever been ceased and desisted lol, still learning the vernacular. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Wednesday, August 12, 2009 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Twitter is Suing me!!! Nowhere does the letter say Twitter is suing you. It is a cease and desist letter. I suppose their threat about further action at the bottom can be reasonably surmised to mean that they might sue you in the future, but that is a far, far cry from Twitter is suing me! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call File Channel
I know I'm missing something here (been a long day)... How can I specify more than one channel in a call file? I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1... Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call File Channel
Thanks Danny, I do have a dial cmd with multiple arguments in my normal outgoing context. I guess my question really is: How do I tell the call file using Channel: XXX to use my outgoing context instead of Zap/g1/xx or sip/trunk_x/xx directly? -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, August 12, 2009 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call File Channel Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Wednesday, August 12, 2009 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call File Channel I know I'm missing something here (been a long day)... How can I specify more than one channel in a call file? I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1... Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call File Channel
Context: is what the call is dumped into after it is answered, at extension Extension:. I don't think it's related to how the call is placed. I can dial the local extension SIP/170 but I'm not sure where that gets me. Basically I want to have the same failover that I have for all other outgoing calls on these automatic calls... Thanks Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, August 12, 2009 5:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call File Channel Ok. Here's how you would do that: Channel: SIP/170 (some local extension) CallerID: SIP/104 (another local extension) MaxRetries: 1 WaitTime: 60 retryTime: 5 Context: your_context Extension: s This should create an extension call using your context. The context can then dial out as you write it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Wednesday, August 12, 2009 4:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call File Channel Thanks Danny, I do have a dial cmd with multiple arguments in my normal outgoing context. I guess my question really is: How do I tell the call file using Channel: XXX to use my outgoing context instead of Zap/g1/xx or sip/trunk_x/xx directly? -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, August 12, 2009 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call File Channel Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Wednesday, August 12, 2009 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call File Channel I know I'm missing something here (been a long day)... How can I specify more than one channel in a call file? I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1... Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call File Channel
Duncan and Danny-- Thank you! I believe the Local/ is what I was missing with ex...@context. -Dave From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Duncan Turnbull [dun...@e-simple.co.nz] Sent: Wednesday, August 12, 2009 5:42 PM To: Asterisk Users Mailing List - Non-Cohttps://mail.videon-central.net/owa/?ae=PreFormActiont=IPM.Notea=Replyid=RgDvdntYewg%2bRopom4XHVQiWBwDABk4e%2fzVQQKMcsNSFUOsuAE10SQAHAAD54%2bBr%2fe7oQrgyh88yX6qLANRp8a4EAAAJ#mmercial Discussion Subject: Re: [asterisk-users] Call File Channel If you use a Local channel to dial it then it will fall under the same rules Channel: Local/numbertod...@the-context-you-want This gets a CDR produced, it does pay to check everything works the same but it should be fine Cheers Duncan David Gibbons wrote: Context: is what the call is dumped into after it is answered, at extension Extension:. I don’t think it’s related to how the call is placed. I can dial the local extension SIP/170 but I’m not sure where that gets me. Basically I want to have the same failover that I have for all other outgoing calls on these automatic calls… Thanks Dave *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, August 12, 2009 5:17 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Call File Channel Ok. Here’s how you would do that: Channel: SIP/170 (some local extension) CallerID: SIP/104 (another local extension) MaxRetries: 1 WaitTime: 60 retryTime: 5 Context: your_context Extension: s This should create an extension call using your context. The context can then dial out as you write it. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Wednesday, August 12, 2009 4:10 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Call File Channel Thanks Danny, I do have a dial cmd with multiple arguments in my normal outgoing context. I guess my question really is: How do I tell the call file using “Channel: XXX” to use my outgoing context instead of Zap/g1/xx or sip/trunk_x/xx directly? -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, August 12, 2009 5:05 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Call File Channel Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60) *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Wednesday, August 12, 2009 3:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call File Channel I know I’m missing something here (been a long day)… How can I specify more than one channel in a call file? I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1… Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com outbound mail slow?
My messages go through rather quickly (minutes). Unless the lists.digium.com server is running on an Atari, it's probably NOT an overload issue... -Dave snip Are there any plans to beef up the mailing list server so that messages can get through with less of a delay? /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CURL function with SSL
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have had
lots of trouble with CURL being able to validate a cert. That's probably
because I didn't tell it where the root certs were... but either way.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wenbin Zhang
Sent: Friday, August 14, 2009 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CURL function with SSL
Tilghman Lesher wrote:
On Friday 14 August 2009 09:04:12 Wenbin Zhang wrote:
Hi all,
I hope you guys can help me out. I got a problem with using function
CURL. I
did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I
generated the call, the CURL function could not get access to that
https://URL server. What should I do with it? Thank you very much
The most likely problem is that your libcurl library was not compiled with
SSL support. If you're installing from source, you'll need to recompile,
adding SSL support to the configure options. If you're installing from a
package, there's likely another package that you need to install for SSL
support (e.g. in Ubuntu, you need to install either libcurl4-gnutls-dev or
libcurl4-openssl-dev).
Thank you very much for your help Tilghman. But I have one more question
here. After I install libcurl4-openssl-dev, do I have to do some
configurations about it? Or do I have to let CURL know the certificate
that https://URL is using? Please tell me some details if you can. Thank
you very much.
___
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Re: [asterisk-users] High Volume Call Center SIP versus IAX2
snip the IAX quality is at best 40-50% of a SIP connection. /snip How is this calculated? Thanks Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, October 20, 2009 4:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] High Volume Call Center SIP versus IAX2 Just my opinion - Call Centers should be SIP trunked because IAX is more prone to poor sound quality. IN MY SHOP (shouting to make the point that I'm not speaking for all Asterisk installations), the IAX quality is at best 40-50% of a SIP connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Tuesday, October 20, 2009 3:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] High Volume Call Center SIP versus IAX2 I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to make sure I am approaching the setup correctly. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
snip What say you to the proposal that some approaches to seeking help are so ridiculous they should not be tolerated? /snip I say give me a break. Pre-judging people doesn't work on mailing lists given the inherent language barriers, etc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing labels on Phones
snip Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14 that uses your ability to press keys on the phone. You could apply the same idea to press the correct buttons to change the background without rebooting. I can't find the script that I found to do this, but I'll keep looking when I get a chance. -Jonathan /snip Here is the snippet of code that I use to set the image on the 79x1 series (should plug right into my reboot script via the link posted by Jonathan): $actions['setimage'][0] = array(0 = Key:Settings, 1 = .3); $actions['setimage'][1] = array(0 = Key:KeyPad1, 1 = .3); $actions['setimage'][2] = array(0 = Key:KeyPad2, 1 = 5); $actions['setimage'][3] = array(0 = Key:KeyPad2, 1 = 3); $actions['setimage'][4] = array(0 = Key:Soft1, 1 = 7); $actions['setimage'][5] = array(0 = Key:Soft2, 1 = 1); $actions['setimage'][6] = array(0 = Key:Soft3, 1 = 1); $actions['setimage'][7] = array(0 = Key:Soft3, 1 = 1); And here is how I set the ringer: $actions['regularring'][0] = array(0 = Key:Settings, 1= .3); $actions['regularring'][1] = array(0 = Key:KeyPad1, 1 = .3); $actions['regularring'][2] = array(0 = Key:KeyPad1, 1 = .3); $actions['regularring'][3] = array(0 = Key:KeyPad1, 1 = 4); $actions['regularring'][4] = array(0 = Key:KeyPad1, 1 = .1); $actions['regularring'][6] = array(0 = Key:Soft1, 1 = 1); $actions['regularring'][7] = array(0 = Key:Soft3, 1 = 1); $actions['regularring'][8] = array(0 = Key:Settings, 1 = .1); I guess I should update my blog with the new version :) *note*: These key sequences are correct on 8-4-1S, I know they have changed in previous firmware changes and chances are they have/will change with versions beyond 8-4-1S. Cheers Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODP: Re: Changing labels on Phones
snip There are some other methods to display content on the phone screen without editing local configs. Check http://www.ciptec.co.uk/ - commercial site but shows the way. /snip If you just want to display user info on the phone, why not use the idle url feature: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2007/09/weather_report_for_idle_screen_on_cisco_ip_phones.html In conjunction with a 15 or 30 second meta refresh tag, this seems like it would be able to pull up-to-date info for the phone display periodically. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 64, Issue 52
snip Not trying to be a smart-a$$, just hoping to find something a little smoother. Is there a better way, or is help as useless as it is starting to appear? /snip If you're actually 'sitting' at the *nix console, use CTRL+PageUP to scroll back up in the buffer. If you ARE using the console directly, it's time to switch to SSH so that you have some scrollback buffer easily available. And so that you don't have to sit next to your * server when you're working on it... Otherwise, use more -or- less to view the output of a * command: `asterisk -r -x help | more` This will paginate and you can scroll through the pages using spacebar. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send the same message to list of users
snip Customers in Europe all have mobile phones, while senders in North America rarely have them ( they have answering machines, though ). /snip What planet/year are you/your clients living on/in? I don't know anyone who doesn't have a mobile. Maybe it's just that they call it a cell phone instead of a mobile :) How could anyone possible consider themselves a serious business person without a cell phone? That's laughable. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7961 - can't place calls
snip Thanks for the reply. I am not getting any output from the Asterisk CLI when I place the call. The phone give busy signal as soon as I push the first digit of the extension #. When I call the 7961 from another extension I get the following on the CLI - that works fine. /snip If the phone gives a fast busy AS SOON as you type a digit, the problem is likely that you need to edit your dialplan.xml file on your TFTP server, so that the phone knows not to send digits immediately after you start typing: Contents of dialplan.xml (customize to fit your situation): DIALTEMPLATE TEMPLATE MATCH=91.. TIMEOUT=0/ TEMPLATE MATCH=9[2-9].. TIMEOUT=0/ TEMPLATE MATCH=10. TIMEOUT=0/ TEMPLATE MATCH=5.. TIMEOUT=0/ TEMPLATE MATCH=605 TIMEOUT=0/ TEMPLATE MATCH=* TIMEOUT=10/ /DIALTEMPLATE -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
I recently implemented a vmware host using SSDs for the VM storage. I wish you could see the grin on my face right now. It's so fast. Remember thought that all SSDs are NOT created equal... Be careful what you buy. snip On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
snip And? Noticed any significant performance advantage? /snip Massive increase in performance on mysql VMs with database sizes that exceed memory size (file caching). Boot times on VMs (windows and linux) under 10 seconds. There is no noticeable change in performance for normal operations on normal VMs because most of the files they're IO blocked by are already cached in memory. I actually went with consumer-grade SSDs (4x OCZ 120gb models) in a raid 10. I know most people say 'those aren't good enough for me'. They are! And as long as you plan for some of them to fail over time, you're still ahead on cost and performance vs enterprise-grade SSDs (read: intel). Synthetic testing with hdparm (sdb is the SSD array, sda is the spinning disk array) is below. This comparison is against 7200rpm disks; I don't have hdparm installed on a box running 15k rpm disks: hdparm -tT --direct /dev/sdb /dev/sdb: Timing O_DIRECT cached reads: 1128 MB in 2.00 seconds = 563.48 MB/sec Timing O_DIRECT disk reads: 1276 MB in 3.00 seconds = 425.01 MB/sec hdparm -tT --direct /dev/sda /dev/sda: Timing O_DIRECT cached reads: 138 MB in 2.03 seconds = 68.03 MB/sec Timing O_DIRECT disk reads: 364 MB in 3.00 seconds = 121.32 MB/sec ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many lines do you use.
I use two 'lines' though 'Line appearances' would be a better term, though still confusing in my book. One line for incoming, one line that auto-answers for paging. Cisco really has so many line appearances on their phones to enable BLF using SIP over TCP. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, November 25, 2009 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How many lines do you use. Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a softphone for testing now. Curious minds are wanting to know ... Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many lines do you use.
snip Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) /snip Are you sure about this? I believe the 79xx series on 8x SIP firmware loads does BLF with SIP/TCP, just not SIP/UDP. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many lines do you use.
snip Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) It does with the skinny firmware :) The skinny channel driver also comes with the 'random crash' feature ;-p. But truth be told I only every tried chan_sccp2 (or was it b...). -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max how many users in sip.conf
snip If you had 1gb of memory, a 200mb load with everything else would be pretty taxing. Hope this is helpful. /snip What distro are you using?? If linux is using 800Mb of memory in an idle state for anything other than file system caching, there's a problem... -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mtha...@gmail.com Sent: Saturday, November 28, 2009 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max how many users in sip.conf Anyone know how many users i can record in sip.conf. (NO..NO i am not discussing the simultaneous sip calls). I tried with 50k users in sip.conf, but the sip module didn't reload. tried with few hundred of users and it works. any idea what is the limit in sip.conf regards Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable Name needed
snip
My question is, Does anyone know what variable I would use to get the
information for To from these SIP calls, the below is the actual SIP packet
obtained from the CLI with SIP Debug On. Other than I stripped out the IPs
/snip
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the SIP headers from callcentric like this:
Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
Which gives me a real US number like 1xx.
Credit for the parsing syntax goes to someone else (not sure where I found it
online).
--Dave
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Re: [asterisk-users] network config
snip A client has two offices in the Virgin Islands that MUST maintain data connectivity, and there are no available leased line options to run a P2P link between them. snip Is there line of sight? I've been wanting to do a long-shot wifi link and my company would give it a shot if you want :). snip Do you lose an in progress call when the tunnel switches from one link to the other? /snip Any 'fail-over' router with links from separate providers that don't route the same subnets (cable/dsl) will have to change its default route when it 'fails-over'. As such, the VPN tunnel will be disconnected and reconnected. I'm sure you could make it brief, but yes, calls will likely be completely dropped. snip And finally - is there a device that will manage the tunnel such that a high water mark of latency will also cause the tunnel to switch to the other link, rather than actual packet loss? /snip See above. Fail-over routers have to wait some criteria are met in order to fail over (ping latency, ping loss, etc). This means that the connection you're using as the 'default' WILL go 'down' BEFORE it switches to the other one, regardless of the criteria used. Another plan would be to set up two routers at the site with two separate VPN tunnels across the two different links, both tunnels being always on. You could then use a SIP proxy or iptables magic to choose which tunnel was the best at any given time. I would go for the wifi. Maybe because I want to do a long-shot link. Also because I want to go to the virgin islands :). Good luck! -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting problem with IP's
snip Just a guess, but the connection probably went from full to half duplex. /snip Full vs. Half duplex networking would NOT cause half duplex phone calls. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple sip trunks
snip
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
/snip
Your provider is probably sending the DID in the SIP header TO: field. This was
discussed on the list last week to at a reasonable level of detail but
generally speaking, you want to dump all of the calls into a context like
[FromSIP] and then have all calls parsed based on the to: field with something
like this:
(credit for this goes to someone at asterisk-info.org, but I didn't write down
who...)
[FromSIP]
;DIDs
exten = 888555,1,Dial(SIP/EXTENSION,10)
;parser
exten = i,1,Goto(FromSIP|s|1)
exten = s,1,Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})
exten = s,n,Goto(FromSIP|${calldest:1}|1)
Then you can set up an exten for each incoming DID that will handle the calls
directly within this same context. Turn on sip debugging and high verbosity at
the cli to help yourself see what's going on with this...
-Dave
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[asterisk-users] IVR Prompt Recording
This may belong on -biz, but does anyone have experience with a decent and cheap IVR/prompt recording house? Are decent and cheap mutually exclusive? A nice *sounding* lady would be nice... you can keep any burly voice studios to yourself :) Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple sip trunks
In that case, you're going to have to talk to your provider. They SHOULD be able to easily send the DID with the call... -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Taylor Sent: Tuesday, December 15, 2009 5:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] multiple sip trunks I thought so- the fact the server has 20 different registry entries to 20 different account all at the same ITSP shouldn't matter? Can't see any DDI info in the SIP headers unfortunately :( John 2009/12/14 meetmecall i...@meetmecall.nlmailto:i...@meetmecall.nl The easiest solution to deal with this is to have one context with different extensions for the different numbers and route the incoming calls from there. It should look something like this (not a tested piece of asterisk script, just an example to give the idea). Hope it helps :-) Erik de Wild [all_trunks] exten = 31592123456,1,Goto(trunk1,s,1) exten = 31592123457,1,Goto(trunk1,s,1) exten = 31592123458,1,Goto(trunk1,s,1) exten = 3159212,1,Goto(trunk2,s,1) exten = 31592123334,1,Goto(trunk2,s,1) exten = 31592123335,1,Goto(trunk2,s,1) On 14 dec 2009, at 10:39, Olle E. Johansson wrote: 11 dec 2009 kl. 23.21 skrev John Taylor: I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? This is an effect of the Asterisk architecture. We've had many discussions on how to change it, but right now the peer matching on IP/Port can't separate various instances from each other, since they all have the same IP/port. Asterisk simply goes for the first match, which happens to be the last entry with the IP/port in the sip.conf file. /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Gmail DOES process those headers... And a proper mail client will also parse the headers and provide unsubscribe information/buttons based on that ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
I haven't had a good mailing list war in a while. Yes, gmail DOES default to top posting, because bottom posting is silly (in general, but especially for a client that hides quoted text (like gmail)). Top posting is modern. And better. And doesn't make me scroll through 10 thousand messages and awful rsa keys to get to the message... FLAME AWAY!!! Press the 'show details' to the right hand side of the message box, then click the link that shows up that says 'unsubscribe'... -Dave snip I use gmail but don't see any buttons for unsubscribe or anything like that? Also, gmail defaults to top posting...which seems to upset some people 'round these parts. I have yet to find a way to make gmail not top-post by default... /snip ___ -- 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
