Re: [asterisk-users] Looking for a way to show caller id information on the desktop
On Windows systems you can use Xtelsio http://www.xtelsio.com/ Jonathan Moore schreef: Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is called to GET the callerid information, but that's not what I'm looking for. Is it possible, and if so, where should I start looking to find a solution to this? I've failed at google so far, and I think I'm just not asking the right question. Thanks for any help or pointers. -jonathan ___ -- 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 -- Met vriendelijke groet Kind Regards, Mit den besten Grüßen, Fons van der Beek, 84-IT BV http://www.84-it.com/index.php?option=com_contentview=articleid=2Itemid=2 T +31 475 769002 M +31 6 29296243 E fons.vanderb...@84-it.com mailto:fons.vanderb...@84-it.com ___ -- 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] Asterisk and Skype
when using sisky you could integrate an ivr menu Alex Balashov schreef: This is not currently possible. Work in progress. -- Sent from mobile device On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval ___ -- 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 -- Met vriendelijke groet Kind Regards, Mit den besten Grüßen, Fons van der Beek, 84-IT BV http://www.84-it.com/index.php?option=com_contentview=articleid=2Itemid=2 T +31 475 769002 M +31 6 29296243 E fons.vanderb...@84-it.com mailto:fons.vanderb...@84-it.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
Re: [asterisk-users] IP door opening devices
Siemens HC 450 Dect intercom does exactly what you want it doesn't come cheap, but works like a dream.. Gordon Henderson schreef: On Thu, 24 Jul 2008, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. There was talk of this a week or 2 ago on the list - look into the archives. I don't think there was anything that successfull though... I have to say though - if you have such an integrated unit that needs nothing more than an analogue connection (and power, presumably), I'd love to know the make - for me, (or rather one of my clients) it would be worthwhile trying to find an ATA that would work with it.. Got a name/website for the opener device? 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] Siemens Gigaset S685IP Review
I also bought the door intercom HC450 it also works like a charm The only drawback: It's not possible (to my knowledge) to make a call TO the HC450-intercom from asterisk The only drawback sofar with C470IP and C475IP are it's not possible to assign ONE handset dedicated to the landline in respect to OUTGOING calls you allways have to pre-dail something. Alan Lord schreef: Marco wrote: Hi Alan, yeah, latest Siemens DECT phones with VoIP support are quite the new Chuck Norris of cordless phones. Personally I use the C470IP on a business context and a C475IP at home (for the integrated answering machine). The audio quality is amazing, and the extra services are definitely a plus. I just found 2 big cons: * The transfer/hold modes is quite a pain, and it takes many keypress to activate, which doesn't make them suitable for all people * The firmware and ALL of the pre-recorded messages are in german. I had some customers a little scared about this! What do you think? Bye Marco Hi Marco, As I've only had them a few days and so haven't used all the features yet. Call transfer is one that I do need to do but, I guess it's the same with most things, once you get into a routine I don't think that 2 or 3 keypresses will seem a chore. As for the language, the phones I bought are for the UK market and everything is English. I have heard of Australians, who seem to have to buy from the UK or Europe getting German or Dutch versions before but it wasn't a problem here. These are UK market units. Ciao Al ___ -- 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] Customer complains of noise on line I cannot reproduce.
Gyboriy, Unfortunatly adding timing=1 didn't help, there are still beeps in the conversations... any other ideas? With kind regards Fons van der Beek Fons van der Beek schreef: tnx for your help Gryboriy, I've added timing=1 to misdn-init.conf Can i check anything to see if it's ok? misdn log gives no errors when loading Thu Mar 6 13:38:55 2008: P[ 4] Got EVENT_FACILITY but we don't have a ch! Sat Mar 8 07:52:34 2008: P[ 0] -- mISDN Channel Driver Registered -- Sun Mar 9 11:49:46 2008: P[ 0] -- mISDN Channel Driver Registered -- Sun Mar 9 12:09:08 2008: P[ 0] -- mISDN Channel Driver Registered -- Shall i wait and see now what is happening? or can i do something else? Grygoriy Dobrovolskyy schreef: Ok lets try same way: add option timing=1 in misdn-init.conf see what happens 2008/3/9, Fons van der Beek [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: I use MiSDN so head -n 1 /proc/zaptel/* gives head: cannot open `/proc/zaptel/*' for reading: No such file or directory Grygoriy Dobrovolskyy schreef: paste output of head -n 1 /proc/zaptel/* 2008/3/9, Fons van der Beek [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: what clock? rxclock crystalclock I currently use card=1,0x4 Grygoriy Dobrovolskyy schreef: Well i have installed asterisk on spare system to replace old one, with new tyan motherboard, surprise came when i installed digium fxs/fxo and b410 p, system unstable, random bips on start, misdn module Not loading, heh, old system worked on asus p5nd2 sli without a problem with them. There were 2 reasons why i wanted change mobo 1: chipset on asus card generated too much heat, 2: i had a new 3ware controller. So you never know really. Loud scrathing sound? sometimes a card problem, try on other hardware. Pci interrupts, also maybe sync problem (you can enable b410 clock in misdn-init.conf) Also turn off all sound/usb/etc unused devices. 2008/3/8, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: [EMAIL PROTECTED] lspci -v -s 01:07.0 pcilib: Cannot open /sys/bus/pci/devices :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ac00 [size=256] Memory at fbfff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 This is all I know about it. The client bought them about a month ago and installed them himself then asked me to setup the Asterisk program for him. The problem motherboard was a four year old Gigabyte with a Promise IDE controller. The new motherboard that works well is an ASUS but I don't know anything else about it. On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which revision of the Digium TDM400? On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27
Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.
tnx for your help Gryboriy, I've added timing=1 to misdn-init.conf Can i check anything to see if it's ok? misdn log gives no errors when loading Thu Mar 6 13:38:55 2008: P[ 4] Got EVENT_FACILITY but we don't have a ch! Sat Mar 8 07:52:34 2008: P[ 0] -- mISDN Channel Driver Registered -- Sun Mar 9 11:49:46 2008: P[ 0] -- mISDN Channel Driver Registered -- Sun Mar 9 12:09:08 2008: P[ 0] -- mISDN Channel Driver Registered -- Shall i wait and see now what is happening? or can i do something else? Grygoriy Dobrovolskyy schreef: Ok lets try same way: add option timing=1 in misdn-init.conf see what happens 2008/3/9, Fons van der Beek [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: I use MiSDN so head -n 1 /proc/zaptel/* gives head: cannot open `/proc/zaptel/*' for reading: No such file or directory Grygoriy Dobrovolskyy schreef: paste output of head -n 1 /proc/zaptel/* 2008/3/9, Fons van der Beek [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: what clock? rxclock crystalclock I currently use card=1,0x4 Grygoriy Dobrovolskyy schreef: Well i have installed asterisk on spare system to replace old one, with new tyan motherboard, surprise came when i installed digium fxs/fxo and b410 p, system unstable, random bips on start, misdn module Not loading, heh, old system worked on asus p5nd2 sli without a problem with them. There were 2 reasons why i wanted change mobo 1: chipset on asus card generated too much heat, 2: i had a new 3ware controller. So you never know really. Loud scrathing sound? sometimes a card problem, try on other hardware. Pci interrupts, also maybe sync problem (you can enable b410 clock in misdn-init.conf) Also turn off all sound/usb/etc unused devices. 2008/3/8, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: [EMAIL PROTECTED] lspci -v -s 01:07.0 pcilib: Cannot open /sys/bus/pci/devices :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ac00 [size=256] Memory at fbfff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 This is all I know about it. The client bought them about a month ago and installed them himself then asked me to setup the Asterisk program for him. The problem motherboard was a four year old Gigabyte with a Promise IDE controller. The new motherboard that works well is an ASUS but I don't know anything else about it. On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which revision of the Digium TDM400? On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL
Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.
what clock? rxclock crystalclock I currently use card=1,0x4 Grygoriy Dobrovolskyy schreef: Well i have installed asterisk on spare system to replace old one, with new tyan motherboard, surprise came when i installed digium fxs/fxo and b410 p, system unstable, random bips on start, misdn module Not loading, heh, old system worked on asus p5nd2 sli without a problem with them. There were 2 reasons why i wanted change mobo 1: chipset on asus card generated too much heat, 2: i had a new 3ware controller. So you never know really. Loud scrathing sound? sometimes a card problem, try on other hardware. Pci interrupts, also maybe sync problem (you can enable b410 clock in misdn-init.conf) Also turn off all sound/usb/etc unused devices. 2008/3/8, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: [EMAIL PROTECTED] lspci -v -s 01:07.0 pcilib: Cannot open /sys/bus/pci/devices :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ac00 [size=256] Memory at fbfff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 This is all I know about it. The client bought them about a month ago and installed them himself then asked me to setup the Asterisk program for him. The problem motherboard was a four year old Gigabyte with a Promise IDE controller. The new motherboard that works well is an ASUS but I don't know anything else about it. On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which revision of the Digium TDM400? On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Check for IRQ issues, move the card to a different slot. You could ask permission to record calls so maybe you can hear the sound yourself. I would then go ahead and swap out cards. I have had TDM400 with bad modules and also bad ports on the cards themselves, so it could a hardware issue. This is what I suspect, especially if you did not put any surge suppression on your telco lines. Usually, at least in my experience, ticks or beeps indicate IRQ, hissing or loud static indicate something with/on the board is bad. ALWAYS use surge suppression on your lines! Thanks, Steve Totaro On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are there known causes for this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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 -- http://www.Radados.org ___ -- Bandwidth and Colocation
Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.
I use MiSDN so head -n 1 /proc/zaptel/* gives head: cannot open `/proc/zaptel/*' for reading: No such file or directory Grygoriy Dobrovolskyy schreef: paste output of head -n 1 /proc/zaptel/* 2008/3/9, Fons van der Beek [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: what clock? rxclock crystalclock I currently use card=1,0x4 Grygoriy Dobrovolskyy schreef: Well i have installed asterisk on spare system to replace old one, with new tyan motherboard, surprise came when i installed digium fxs/fxo and b410 p, system unstable, random bips on start, misdn module Not loading, heh, old system worked on asus p5nd2 sli without a problem with them. There were 2 reasons why i wanted change mobo 1: chipset on asus card generated too much heat, 2: i had a new 3ware controller. So you never know really. Loud scrathing sound? sometimes a card problem, try on other hardware. Pci interrupts, also maybe sync problem (you can enable b410 clock in misdn-init.conf) Also turn off all sound/usb/etc unused devices. 2008/3/8, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: [EMAIL PROTECTED] lspci -v -s 01:07.0 pcilib: Cannot open /sys/bus/pci/devices :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ac00 [size=256] Memory at fbfff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 This is all I know about it. The client bought them about a month ago and installed them himself then asked me to setup the Asterisk program for him. The problem motherboard was a four year old Gigabyte with a Promise IDE controller. The new motherboard that works well is an ASUS but I don't know anything else about it. On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which revision of the Digium TDM400? On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Check for IRQ issues, move the card to a different slot. You could ask permission to record calls so maybe you can hear the sound yourself. I would then go ahead and swap out cards. I have had TDM400 with bad modules and also bad ports on the cards themselves, so it could a hardware issue. This is what I suspect, especially if you did not put any surge suppression on your telco lines. Usually, at least in my experience, ticks or beeps indicate IRQ, hissing or loud static indicate something with/on the board is bad. ALWAYS use surge suppression on your lines! Thanks, Steve Totaro On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound
Re: [asterisk-users] sip show channels - gives a growing list of dead channels
Same problem over here I use KIRK-Telecom ip600v3 This only happens on calls between SIP en MiSDN, anyone any clue? As far as i can see these dead calls once in while occur when the remote party first hangs up (remote=MiSDN channel) Keith do you also have error messages in the CLI when you open asterisk by using asterisk -rvv ? (a lot of v) -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 10.0.0.71 represents the IP number of internal phone Keith Hardee schreef: I am using Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 7fa08c964a1 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 542 7088c6a7-db 00102/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 424cc109052 00104/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 541 225fe5130e5 00104/00102 0x0 (nothing) No Rx: BYE Thanks, Keith ___ -- 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] Digium TDMXXB and Electronic Noises
Did interupt sharing caused this problem or are you still having this problem? After i checked and solved the IRQ sharing I still have this problem. I use an TDM410B. I use misdn Volume settings are default Matthew Yingling schreef: I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling ___ -- 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] Customer complains of noise on line I cannot reproduce.
Perhaps irq sharing? Royce Souther schreef: I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are there known causes for this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] Music on hold
I guess we are back to the fundamental problem: no asterisk generated sounds on the external call After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Sip show peers for the other end shows: * Name : sip.xs4all.nl Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : default Language : en AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 0475769XXX FromDomain : sip.xs4all.nl Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sip.xs4all.nl Addr-IP : 82.101.XX.XX Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 0475769XXX SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : Trevor Peirce schreef: Fons van der Beek wrote: I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? A simple way for you to test your indications.conf as far as the ringing goes is something like this: exten = s,1,Answer exten = s,n,PlayTones(ring) exten = s,n,Wait(30) exten = s,n,Hangup That should pick up the line and then play your locale's ring tone for 30 seconds before hanging up. If you hear ringing then indications.conf is fine, otherwise you have confirmed that there is a problem somewhere. This will have nothing to do with your carrier as the sounds are generated by asterisk itself as audio (as opposed to any kind of carrier-specific signaling). Trevor Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- 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] Music on hold
While the call is progressing sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 82.101.62.XX 0475769XXX 14151-EX-29 00101/703757593 0x4 (ulaw) No Rx: ACK 82.101.62.XX 0475769XXX 6ec6f62d57d 00103/0 0x0 (nothing)No Codec=Ulaw, still no ringing Fons van der Beek schreef: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Sip show peers for the other end shows: * Name : sip.xs4all.nl Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : default Language : en AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 0475769XXX FromDomain : sip.xs4all.nl Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sip.xs4all.nl Addr-IP : 82.101.XX.XX Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 0475769XXX SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : Trevor Peirce schreef: Fons van der Beek wrote: I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? A simple way for you to test your indications.conf as far as the ringing goes is something like this: exten = s,1,Answer exten = s,n,PlayTones(ring) exten = s,n,Wait(30) exten = s,n,Hangup That should pick up the line and then play your locale's ring tone for 30 seconds before hanging up. If you hear ringing then indications.conf is fine, otherwise you have confirmed that there is a problem somewhere. This will have nothing to do with your carrier as the sounds are generated by asterisk itself as audio (as opposed to any kind of carrier-specific signaling). Trevor Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- 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] Music on hold
Tnx for your support Trevor!! cat /etc/asterisk/indications.conf | grep country= country=nl ; default location show indications nl Country Indication PlayList = nl ringcadence 1000,4000 nl dial425 nl busy425/500,0/500 nl ring425/1000,0/4000 nl congestion 425/250,0/250 nl callwaiting 425/500,0/9500 nl dialrecall 425/500,0/50 nl record 1400/500,0/15000 nl info950/330,1400/330,1800/330,0/1000 nl stutter 425/500,0/50 The 'show indications' command is deprecated and will be removed in a future release. Please use 'indication show' instead. But Trevor, I guess this isn't the problem, because when i call from an internal location the indication is all right Also moh works from internal SIP phones to the queue. I only have a problem when i call into my asterisk box from the outside. Trevor Peirce schreef: Fons van der Beek wrote: After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Alright, well let's see what ring actually is set to for your system. Let's see this from the command line: cat /etc/asterisk/indications.conf | grep country= And this from asterisk: show indications XX (where XX is your locale, of course). ___ -- 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] Music on hold
Jared YES That seems to be the problem! A very very long time ago I installed a X101P (an original one) and forgot about it. After issuing a modprobe ztdummy, indications on the outside line indication work as they should. After that i configured my X101P the way it should be configured! And yes! now indications are the way they should be I rebooted, I restarted asterisk and it keeps working! I want to thank everyone who helped me, Thank you all Jared Smith schreef: On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call Do you have any T1/E1 cards in your system that aren't configured? If a zaptel card isn't taking interrupts, that would cause this same type of problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
but your support was superior Eric! tnx for your help! Eric Wieling schreef: I must have started reading this thread after you reported that you actually had an AUDIO problem rather than a RINGBACK problem. The issue you experienced is a common one. Someday I hope Digium fixes that bug/design flaw. Fons van der Beek wrote: Jared YES That seems to be the problem! A very very long time ago I installed a X101P (an original one) and forgot about it. After issuing a modprobe ztdummy, indications on the outside line indication work as they should. After that i configured my X101P the way it should be configured! And yes! now indications are the way they should be I rebooted, I restarted asterisk and it keeps working! I want to thank everyone who helped me, Thank you all Jared Smith schreef: On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call Do you have any T1/E1 cards in your system that aren't configured? If a zaptel card isn't taking interrupts, that would cause this same type of problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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] Music on hold
Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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] Music on hold
it's very odd -I just upgraded to 1.4.18 (from 1.4.17) -removed answer -changed to several other options, still no luck (restarted also) Eric Wieling schreef: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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] Music on hold
NOT answering did the trick! Tnx a lot! now it works like it should work! Eric Wieling schreef: Replying to my own post. Asterisk uses indications.conf when it has to provide tones AFTER the line is answered. You might get a message on the console like Unable to handle indication 15 or something like that. Eric Wieling wrote: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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] Music on hold
I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? and even with verbose set to 114 (quite big) there are no errormessages indicating that something is wrong with indications (in respect to syntax) Eric Wieling schreef: If not answering fixes the problem then the issue is indications.conf. Try using the indications.conf.sample file included with the Asterisk source code, then stop Asterisk and starting it again. I do not know if indications.conf is reloaded on a reload. Fons van der Beek wrote: NOT answering did the trick! Tnx a lot! now it works like it should work! Eric Wieling schreef: Replying to my own post. Asterisk uses indications.conf when it has to provide tones AFTER the line is answered. You might get a message on the console like Unable to handle indication 15 or something like that. Eric Wieling wrote: Don't answer the line. Also try using the US indications, just in case something odd is in the NL setup. Fons van der Beek wrote: Tnx. I checked /etc/asterisk/indications.conf and my default location nl is listed in the options So i am still puzzled my extensions.conf in respect to incomming calls (as basic as possible) exten = s,1,Answer exten = s,2,queue(receptie|r) exten = s,3,Voicemail(201) everything else works as it should work, but no ringing on an external line on the other hand, internaly: it's ok exten = 205,1,queue(receptie|r) exten = 205,2,busy 205 gives ringing Eric Wieling schreef: This problem would happen if you did not have /etc/asterisk/indications.conf Fons van der Beek wrote: I tried that, its gives me the same problem. Kevin P. Fleming schreef: Fons van der Beek wrote: Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok The Queue() application has an option to generate ringback to callers instead of music on hold, why don't you just use that instead of trying to craft a new solution? ___ -- 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
[asterisk-users] Music on hold
Because i want a ringing signal while people are in a waiting queue i've
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, but
when i make an external call, no signal is heard.
everything else looks ok, and all other functions are ok
Can somebody point me out what i could have done wrong?
my musiconhold.conf
---
[ringing]
mode=files
directory=/var/lib/asterisk/ringing
i've created the wav file using
sox ringing.wav -r 8000 -c 1 -s -w ringingmono.wav resample -ql
This file is located at /var/lib/asterisk/ringing
my queues.conf
[receptie]
musicclass = ringing
strategy = ringall
timeout = 300
retry = 5
member = SIP/202
member = SIP/227
the CLI shows..
---
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/04757690XX-08ef8ee8, ) in
new stack
-- Executing [EMAIL PROTECTED]:2] LookupCIDName(SIP/0475769002-08ef8ee8,
) in new stack
[Feb 17 13:18:10] WARNING[28267]: app_lookupcidname.c:72
lookupcidname_exec: LookupCIDName is deprecated. Please use
${DB(cidname/${CALLERID(num)})} instead.
-- Executing [EMAIL PROTECTED]:3] Queue(SIP/04757690XX-08ef8ee8,
receptie) in new stack
-- Started music on hold, class 'ringing', on SIP/04757690XX-08ef8ee8
-- SIP/227-08efce58 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/414-08f04930 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/201-08f06f68 is ringing
-- SIP/201-08f06f68 answered SIP/04757690XX-08ef8ee8
-- Stopped music on hold on SIP/0475769002-08ef8ee8
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Re: [asterisk-users] message: !! Got Busy in Connected State !?!
What phone do you use? Linksys ? Vieri schreef: --- Fons van der Beek [EMAIL PROTECTED] wrote: Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call off hold And then the external line is disconnected. I get the same with Asterisk 1.2 and chan_misdn. Is this a known bug or something I misconfigured? In the latter case, what should I look for? Thanks, Vieri Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] message: !! Got Busy in Connected State !?!
Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call off hold And then the external line is disconnected. an exact log of events is recorded and is given below: Can somebody give me a clue how to solve this issue ?? [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Executing [EMAIL PROTECTED]:1] Answer(mISDN/5-u299, ) in new stack [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Executing [EMAIL PROTECTED]:2] LookupCIDName(mISDN/5-u299, ) in new stack [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Changed Caller*ID name to some-one,Firstname [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Executing [EMAIL PROTECTED]:3] GotoIf(mISDN/5-u299, 0?7) in new stack [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Executing [EMAIL PROTECTED]:4] Devstate(mISDN/5-u299, 100|1) in new stack [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Executing [EMAIL PROTECTED]:5] Dial(mISDN/5-u299, SIP/201SIP/240|45|twk) in new stack [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Called 201 [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- Called 240 [2008-02-11 16:02:27] VERBOSE[6543] logger.c: Extension Changed 240 new state Ringing for Notify User 201 [2008-02-11 16:02:27] VERBOSE[22205] logger.c: -- SIP/201-08758540 is ringing [2008-02-11 16:02:28] VERBOSE[22205] logger.c: -- SIP/240-b630ef30 is ringing [2008-02-11 16:02:33] VERBOSE[22205] logger.c: -- SIP/201-08758540 answered mISDN/5-u299 [2008-02-11 16:02:33] VERBOSE[6543] logger.c: Extension Changed 240 new state Idle for Notify User 201 [2008-02-11 16:02:40] VERBOSE[22205] logger.c: -- Started music on hold, class 'default', on mISDN/5-u299 [2008-02-11 16:02:42] VERBOSE[22215] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/201-0872f2c0, SIP/211|20|tWwTkK) in new stack [2008-02-11 16:02:42] VERBOSE[6543] logger.c: Extension Changed 211 new state Ringing for Notify User 201 [2008-02-11 16:02:42] VERBOSE[22215] logger.c: -- Called 211 [2008-02-11 16:02:42] VERBOSE[22215] logger.c: -- SIP/211-0874d9a0 is ringing [2008-02-11 16:02:55] VERBOSE[6545] logger.c: -- Stopped music on hold on mISDN/5-u299 [2008-02-11 16:02:55] VERBOSE[22205] logger.c: == Spawn extension (incomming, 0475426426, 5) exited non-zero on 'SIP/201-0872f2c0ZOMBIE' [2008-02-11 16:03:00] VERBOSE[22217] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/201-08775598, SIP/211|20|tWwTkK) in new stack [2008-02-11 16:03:00] VERBOSE[22217] logger.c: -- Called 211 [2008-02-11 16:03:00] VERBOSE[22217] logger.c: -- SIP/211-08779510 is ringing [2008-02-11 16:03:02] VERBOSE[22215] logger.c: -- Nobody picked up in 2 ms [2008-02-11 16:03:02] VERBOSE[22215] logger.c: -- Executing [EMAIL PROTECTED]:2] Busy(mISDN/5-u299, ) in new stack [2008-02-11 16:03:02] WARNING[22215] chan_misdn.c: -- !! Got Busy in Connected State !?! ast:mISDN/5-u299 [2008-02-11 16:03:03] VERBOSE[6543] logger.c: Extension Changed 211 new state Idle for Notify User 201 [2008-02-11 16:03:03] VERBOSE[22217] logger.c: == Spawn extension (default, 211, 1) exited non-zero on 'SIP/201-08775598' ___ -- 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] calls get stuck in the asterisk box
At the end of the day SIP calles keep stuck in asterisk, is there any way to prevent this or debug this? The sip calls which get stuck all are calles on a krik IP600v3 dect gateway, I cant tell if they originate of the ip600v3, probably this are calls TO the IP600v3 10.0.0.71240 2c2cfcc47ca 05593/103700 0x0 (nothing) No Tx: BYE Done 10.0.0.71238 d4b2f570e90 00105/103150 0x0 (nothing) No Rx: BYE 10.0.0.71240 5d02b0d503e 06353/102998 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 4b303fed159 16797/93872 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 181151d9010 16819/93839 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 4abf61ec5ee 18318/92482 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 43a74c2f08d 19014/91859 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 672a3a624b5 19237/91616 0x0 (nothing) No Tx: BYE Done 10.0.0.71240 4ede9bb258e 19332/91525 0x0 (nothing) No Tx: BYE Done 9 active SIP channels -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 -- Incoming call: Got SIP response 400 Bad Request back from 10.0.0.71 the sip.conf for the phones on the IP600v3 all have this settings in sip.conf [239] type=friend username = 239 callerid=name 239 host = dynamic secret = 239 context = default qualify = yes login = 239 callgroup = 3 pickupgroup = 3 disallow = all allow = alaw call-limit = 6 setting of call-limit to 1 doesn't prevent the above mentioned problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] autoprovision 200+ linksys phones setup
I used this as a manual for blukprovisioning.. http://voipspeak.net/index.php?option=com_contenttask=viewid=73Itemid=28 Rob Hillis schreef: Hi Eric, You may want to contact me off-list - the company I work for offers a product which aims to be a zero configuration service for Asterisk. The Linksys 942 and 962 phones /are/ supported. Erick Perez wrote: Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for us but im sure there must be something we can do to speed things up. We are looking into a distribution (freepbx or pure asterisk,or something else) with links to documentation to enable autoprovisioning on the linksys phones. What we want to achieve is enabling the linksys phones to be plugged into the lan, grab a configuration from tftp or http and be assigned the next free extension. (fonality does something like that with polycoms) So far, the autoprovisioning links i've found talk about polycom phones and grandstream. but in this office (and country) linksys is better to get and much less expensive than polycom phones. Maybe some distro i haven't checked out that autoprovisions linksys 942? also, a guidance (howto, manual, web link) on autoprovisioning will be gladly welcomed. 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 ___ -- 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] calls get stuck in asterisk
When there are 3 or more incomming calls on my Linksys 962, it often happens that the calls are in the wait-state but not visible on the phone anymore This happens when we want to transfer the call but the remote party doesn't answer. When we want to take the call back it is gone, but the calling party still has music on hold after a while this times out, and the call is ended by asterisk Does this sound familair to anyone? How can this be debugged? what information does anyone need to help me with this problem? With kind regards Fons van der Beek ___ -- 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] Calls Being Randomly Bridged
Tilghman Lesher schreef: On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. I had exact the same problem in using the snom 360, it's too easy to bridge 2 calls, it isn't a bug, it works as designed but transfering a call on a 360 isn't as user friendly as it should be, specially when many calls are incoming. I've replaced the snom 360 by a linksys 962 and disabled blind transfer. But be warned. When using the 962 and the extra panel train you users using the numeric keypad when transfering calls, using the extra buttonpanel when transferring calls randomly results in loosing calls. Personally i'am still looking for a good station when a lot of incoming trafic is on a main station. ___ -- 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] Calls Being Randomly Bridged
Michael J. Liberatore schreef: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fons van der Beek Sent: Sunday, January 20, 2008 3:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Being Randomly Bridged Tilghman Lesher schreef: On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. I had exact the same problem in using the snom 360, it's too easy to bridge 2 calls, it isn't a bug, it works as designed but transfering a call on a 360 isn't as user friendly as it should be, specially when many calls are incoming. I've replaced the snom 360 by a linksys 962 and disabled blind transfer. But be warned. When using the 962 and the extra panel train you users using the numeric keypad when transfering calls, using the extra buttonpanel when transferring calls randomly results in loosing calls. Personally i'am still looking for a good station when a lot of incoming trafic is on a main station. I think this is the cause too. I checked the logs for parking to direct spots and I didn't see any of that going on so I think this is the likely cause. I disabled the conference button but I think the problem is with transfers as you mentioned. Can anyone think of a way to prevent connecting two callers with the transfer function? Either in the phone or asterisk? I need to have the ability to transfer, but NEVER connect two incoming callers, only connect an incoming caller with a different internal phone. How do you think 2 outside callers are getting bridged with transfering? Thanks Mike Also to the person asking for more detail logs, I will try to get them, they can never tell me exactly when this happens only that it happened a bunch of times this week On the snom 360 If you pay close attention when you transfer the calls, you can see the names/numbers of the calling partners by using the cursor button (the round button with arrows) you can select to who you want to transfer to. It's an user issue, but you can't blame the user when there is a lot incoming traffic it takes too many button presses and careful attention to make a correct transfer. How to disable it? I don't know but i faced the problem that users occasionally want to bridge calls. e.g. someone calls for a person that only can be reached by Cellphone, this can be accomplished by asterisk and is often needed. Personally I'm still looking for a good solution for a central station that is easy to use and has a professional appeal, i thought the linksys 962+932 was it, but it has also some drawbacks. One(or two) button attended transfer is not reliable. certainly not when there are 2 or three simultaneously incoming calls. It gets confusing at that time. If anyone has any suggestions don't hesitate to make them! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] blf and misdn
Fons van der Beek schreef: Hello Is het possible to assign blf to a misdn channel? I want to watch the status of my external misdn channels on a linksys 962, e.g. green = available , red = in use and as an extra I want when I press the blf use the external line or when busy i want to barge in that call. by the way, I'm using asterisk 1.4.17 and devstate Did somebody do this before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] blf and misdn
Hello Is het possible to assign blf to a misdn channel? I want to watch the status of my external misdn channels on a linksys 962, e.g. green = available , red = in use and as an extra I want when I press the blf use the external line or when busy i want to barge in that call. Did somebody do this before? ___ -- 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] Kirk and asterisk
Hello all, I know it was on the list before but i have some questions about the Kirk IP600v3, the requested configuration files were send private i guess Does anybody have the correct SIP settings for handsets connected to the Kirk. IP600v3 I am particulair intrested in settings regarding: -Voice Mailbox -Call waiting -DTMF settings for e.g. parking an extension with asterisk functionality Lately i'am having also trouble when i initiated a transfer, i can't take back the call by pressing R. Does anybody use relaxdtmf and or special DTMF timings for correct usage of the kirk 600v3 ? I am using asterisk 1.4.14 and the newest firmware of the Kirk (07-60663 ) When enabling all advanced features of the kirk 600v3 occasionaly handsets get disconnected, still trying to figure out which of those features create this disconnection. When using no features connection to all handsets are stable. Futhermore i am getting an error on my CLI Incoming call: Got SIP response 400 Bad Request back from 10.0.0.XX when looking in set debug ip to my wireless server SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.0.0.70:5060;branch=z9hG4bK42bce6b1;rport From: XX sip:[EMAIL PROTECTED];tag=as7a91af96 To: sip:[EMAIL PROTECTED]:16406;user=phone;tag=2870354154 http://www.snapanumber.com/ Call-ID: [EMAIL PROTECTED] CSeq: 14806 BYE Server: (KIRK Wireless Server 600v3/6.00 dvl-sr2 [07-60663]) XX = caller id of Calling party It looks OK, but is giving a Bad request Does anybody know how to avoid/solve this error, i get a lot of them Sip.conf for a particulair handset [235] type=friend username = 235 callerid=R Vermeeren mobiel 235 host = dynamic secret = 235 context = default qualify = yes login = 235 callgroup = 3 pickupgroup = 3 disallow = all allow = alaw call-limit = 6 default section of sip.conf [general] dtmfmode=rfc2833 rfc2833compensate=yes notifyringing=yes context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers = yes ; Apply call limits on peers only. This will improve useclientcode=yes When more information is needed, please ask.. ___ -- 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] 7970 CTLFile.tlv?
Try to change your verbose setting of tftboot server and look what file is asked for exactly Matthew Rubenstein schreef: I did (/tftpboot/CTLFile.tlv), but the phone keeps complaining (no change). On Fri, 2007-12-21 at 10:25 -0500, Anciso, Roy wrote: I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv? I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin ___ --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] Linksys SPA962 with SPA932 unexpected reboots
Gaps is not a problem I also have them and have no reboot problems Marty Mastera schreef: Paul, Thanks for your response. I saw the 5.2.2 firmware as well. I might try it, but I was told that this issue was fixed many releases ago. One other thought, I mentioned that the format of my 932 config is: fnc=blf+sd;[EMAIL PROTECTED] which I think is proper, but one somewhat abnormal thing is that I'm not using the 932's slots in order. I have 1-16 configured, then 20-23 (because I was going in order of the extensions and we don't currently have extensions 17-19). Maybe this gap in the configuration is causing problems? Does anyone else out there have gaps in their 932 button usage? Thanks, marty There is a 5.2.2 firmware available now, but the changelog for it isn't helpful at all. PaulH On Wed, 2007-12-12 at 11:22 -0700, Marty Mastera wrote: We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly – could be onhook, could be on a call, doesn’t seem to matter. I read that certain early firmware revisions could cause this so I’m running what was a week ago the newest available (5.1.18). A call to Linksys support suggested that I ensure that the phones are using a recent firmware version (there’s one newer release just a few days ago) and to disable provisioning via the web interface to see if that would help. Any advice from users of this phone/sidecar out there? Details: - SPA962 w/ v5.1.18 firmware and SPA932 sidecar - The format of my SPA932 config is: fnc=blf +sd;[EMAIL PROTECTED] - Using trixbox with Asterisk 1.2.24 I fully expected the newer firmware release that I’m using to prevent this based on user reports from prior firmware versions and I’m unsure of the validity of disabling provisioning since googling the issue didn’t produce any confirmation that anyone else has had to do this… Thanks marty No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.1/1182 - Release Date: 12/12/2007 11:29 AM ___ --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] Linksys SPA962 with SPA932 unexpected reboots
Try to use a static IP adress Perhaps trubble on your dhcp server? Marty Mastera schreef: We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly – could be onhook, could be on a call, doesn’t seem to matter. I've had no problems at all with my SPA-962/932 combo, and I've used all kinds of different firmware versions. If I had to guess, I'd guess that it is a bad phone or bad power adapter. --- Jared Smith Community Relations Manager Digium, Inc. ___ Thanks for the input. I agree with the thought of bad hardware at first glance, but it's happening to all 962/932 combos on this site (3 total). As for power, the phones are using POE which I've considered as an outside possibility for the problem (possibly due to higher power requirements with the attached 932?). Thanks again and keep the suggestions coming. marty No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.1/1182 - Release Date: 12/12/2007 11:29 AM ___ --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 Hipath interconnections
If Siemens claims it is Open source, they also should provide the download link for the software...otherwise it wouldn't be OPEN source - Original Message - From: Tele Cost Price Reducer To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 27, 2006 10:25 AM Subject: Re: [Asterisk-Users] Asterisk and Hipath interconnections hi all, maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system. why am i saying this? because Siemens announce it is a Linux, Open Source system. so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something else? otherwise, if it is an Asterisk system, so why there is a need for Cornet? you can interconnect with IAX, isn't it? Mickey On 2/27/06, Stephen Arulraj [EMAIL PROTECTED] wrote: Hi VictorLooking for the same answers here too. We are regional distributors forHicom HiPath in this part of the world and until now we are still waiting for chan_cornet to come around. So far we have successfullyinterconnected via BRI (mISDN) and PRI (Zaptel) and it works great.Let's see if it's too good to be true soon.Best regards,Stephen Viktor Tatianin wrote:HelloCan anyone know where may download chan_cornet for interconnection Asteriskand Hipath via IPThanksViktor___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
