[asterisk-users] Call to an arbitrary outbound number by asterisk
Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the voicemail is running). I cannot understand why asterisk is doing this whereas my dialplan says it needs to connect to other number exten = _.,1,Dial(Zap/g1/19173995791) Also any idea if this is an Asterisk problem or a telco problem. Any help/hints/suggestions would be most welcome Here are my files. zapata.conf context=incoming switchtype=national signalling=pri_cpe group=1 channel=1-23 extension.conf [incoming] exten = _.,1,Dial(Zap/g1/19173995791) # I have added this line in the dialplan is because I want it to match the last 5 digit and simply dial the number 19173995791 such that a call leg is established between the calling party and the number 19173995791 CLI debug information -- Requested transfer capability: 0x00 - SPEECH -- Called g1/19173995791 -- Zap/1-1 is proceeding passing it to Zap/23-1 -- Zap/1-1 is making progress passing it to Zap/23-1 ### The call keeps ringing for sometime then it goes to voicemail. The message comes when the voicemail start. Note that I have not setup any voice mail -- Zap/1-1 answered Zap/23-1 ### Goes to the voicemail -- Native bridging Zap/23-1 and Zap/1-1 -- Channel 0/23, span 1 got hangup request -- Hungup 'Zap/1-1' == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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
Re: [asterisk-users] HPEC audio clipping
Eric ManxPower Wieling wrote: What are the advantages of 9.x over the 8.x that I currently use? I was about to ask the same question. What if my 8.x EC works just fine? (Why expose yourself to the possibility that even the patched version fails?) -Stephen- ___ --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
RE: [asterisk-users] Call to an arbitrary outbound number by asterisk
From: Arpit Mehta [EMAIL PROTECTED] Date: Fri, 18 May 2007 02:31:22 -0400 Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the voicemail is running). I cannot understand why asterisk is doing this whereas my dialplan says it needs to connect to other number exten = _.,1,Dial(Zap/g1/19173995791) Also any idea if this is an Asterisk problem or a telco problem. Any help/hints/suggestions would be most welcome If you are sure that your university doesn't have a PBX, that's a telco problem. Looks like that the switch has a dial plan that does not allow you to dial this sequence directly and interpret all dialed sequence as a local call. (This is usually the function of a PBX but ...) What is this number 19173995791, any way? (and what is 212-85?) If you attach a phone directly to a channel bank, would you be able to dial this sequence? Yuan Liu Here are my files. zapata.conf context=incoming switchtype=national signalling=pri_cpe group=1 channel=1-23 extension.conf [incoming] exten = _.,1,Dial(Zap/g1/19173995791) # I have added this line in the dialplan is because I want it to match the last 5 digit and simply dial the number 19173995791 such that a call leg is established between the calling party and the number 19173995791 CLI debug information -- Requested transfer capability: 0x00 - SPEECH -- Called g1/19173995791 -- Zap/1-1 is proceeding passing it to Zap/23-1 -- Zap/1-1 is making progress passing it to Zap/23-1 ### The call keeps ringing for sometime then it goes to voicemail. The message comes when the voicemail start. Note that I have not setup any voice mail -- Zap/1-1 answered Zap/23-1 ### Goes to the voicemail -- Native bridging Zap/23-1 and Zap/1-1 -- Channel 0/23, span 1 got hangup request -- Hungup 'Zap/1-1' == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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
Re: [asterisk-users] Outside lines are just not happening...
Stephen Bosch wrote: If you get dumbfounded responses ask to speak to someone in the programming group (unless they are a tiny little phone company, they will have one). If you open a ticket, it usually means they will escalate the problem, even if the agent you are speaking with has no idea what you are talking about. Best to be friendly with them! And what do you do when they say: We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye. And then the call ends. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quadbri Cellular Issue
These are empty files. I've tried deleting this final lines and there's no change... _ Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349 ___ --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
Re: [asterisk-users] Outside lines are just not happening...
Brian Capouch wrote: Stephen Bosch wrote: If you get dumbfounded responses ask to speak to someone in the programming group (unless they are a tiny little phone company, they will have one). If you open a ticket, it usually means they will escalate the problem, even if the agent you are speaking with has no idea what you are talking about. Best to be friendly with them! And what do you do when they say: We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye. Ask for the supervisor. Don't quit pestering them until you get someone who can help you. DON'T QUIT. You'd be surprised what persistence will get you (that is, more than just disconnect supervision) -Stephen- ___ --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
Re: [asterisk-users] Call to an arbitrary outbound number by asterisk
Hi, hi, 19173995791 is some number which I want to dial. 212-85- all/most of the numbers in my workplace start with this - so I presume it has got to do something with this. thanks for your suggestions regards arpit On 5/18/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Arpit Mehta [EMAIL PROTECTED] Date: Fri, 18 May 2007 02:31:22 -0400 Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the voicemail is running). I cannot understand why asterisk is doing this whereas my dialplan says it needs to connect to other number exten = _.,1,Dial(Zap/g1/19173995791) Also any idea if this is an Asterisk problem or a telco problem. Any help/hints/suggestions would be most welcome If you are sure that your university doesn't have a PBX, that's a telco problem. Looks like that the switch has a dial plan that does not allow you to dial this sequence directly and interpret all dialed sequence as a local call. (This is usually the function of a PBX but ...) What is this number 19173995791, any way? (and what is 212-85?) If you attach a phone directly to a channel bank, would you be able to dial this sequence? Yuan Liu Here are my files. zapata.conf context=incoming switchtype=national signalling=pri_cpe group=1 channel=1-23 extension.conf [incoming] exten = _.,1,Dial(Zap/g1/19173995791) # I have added this line in the dialplan is because I want it to match the last 5 digit and simply dial the number 19173995791 such that a call leg is established between the calling party and the number 19173995791 CLI debug information -- Requested transfer capability: 0x00 - SPEECH -- Called g1/19173995791 -- Zap/1-1 is proceeding passing it to Zap/23-1 -- Zap/1-1 is making progress passing it to Zap/23-1 ### The call keeps ringing for sometime then it goes to voicemail. The message comes when the voicemail start. Note that I have not setup any voice mail -- Zap/1-1 answered Zap/23-1 ### Goes to the voicemail -- Native bridging Zap/23-1 and Zap/1-1 -- Channel 0/23, span 1 got hangup request -- Hungup 'Zap/1-1' == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --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
Re: [asterisk-users] how to define a key to decline incoming call
Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]: Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a SIP/2.0 486 Busy Here message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the call. Acting on feature.conf we succed in defining keys for blind transfer or attended transfer: the last thing we need is the ability to drop an incoming call without answering it. Is there any way to define a key (or double-key, i.e. *4) to send back a SIP/2.0 486 Busy Here message ? thanks in advance, Please have a look at http://www.voip-info.org/wiki/view/Asterisk+Cmd+Dial Especially the M() parameter. There is an example 2 that needs only little changing to match your idea. You could, for example, wait 2 secs in that macro before bridging the calls - and reject call if 1 is pressed within 2 seconds, or similar. Of course this prepends 2 seconds to any call bridging. For the Siemens Gigasets, most of them (and I do not know the C450IP, only the C450) show a soft button Ignore which meens stop ringing, but do not tell the caller. Combined with a Dial() timeout and following voicemail this works like stop disturbing, eventually the voicemail will take the call. No idea about the ATCOM though, so having that Macro stuff might be the most universal method available. BR Anselm ___ --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
RE: [asterisk-users] zaptel huge irq problem
I'm no Asterisk expert (nor Linux expert) but I am using my * box for multiple things (transparent firewall, NAT box, samba server, poptop server) and for a considerable time I've been running a VmWare server with a Windows XP virtual machine up-and-running at all times! The Windows XP VM was running IIS, Apache, WarFTP and a Firebird database server - all of which got moderate use. The hardware for my * box is what would be considered moderate-to-cheap: Sempron-something processor (not a big processor, don't remember the exact GHz), enough RAM (I've added 1 Gb of RAM when I've started using VmWare server), a nice motherboard (I remember I specifically looked for a motherboard with the minimum amount of on-board devices, of which I have disabled everything I don't need!). The extra hardware on my box includes 2 PCI NIC's (I'm also using the on-board NIC so I've got 3 working NIC's), an TDM400 card with 3 FXO and 1 FXS, and an Diva Eicon Server BRI card for my ISDN connection. I've got 3 HDD's into the box, of which 2 are old IDE drivers (parallel ATA) and the other one is SATA. My VoIP experience has been good, my zaptel timing is pretty good and I can get faxes working on the FXS interface as well (coming in over the ISDN line). The rationale behind placing the Windows XP virtual machine on the * has not been the lack of extra hardware but the desire to keep the number of always-on servers to a minimum. I've since moved the VM off the Asterisk server because I've installed an Windows SBS 2003 server on a considerably more powerful server. So there it goes, proof that a small-office Asterisk box can do lots and lots of things! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Thursday, May 17, 2007 8:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] zaptel huge irq problem Hi, Why are you so determined to use Asterisk in a VM? You're asking for trouble. Asterisk belongs on dedicated hardware. I actually want to use Asterisk in a machine HOSTING a VM (that's what I implied with the Dom-0 thing I said earlier), sorry for the misunderstanding. I agree with you that given the state of advancement of just about any 'virtualizer', I would have to be totally stupid to try running Asterisk inside a VM. (I also wouldn't have asked here in the first place, as I would have been totally certain that problems came from the virtualizer itself) If you feel concerned with my reasons for doing that anyway: - No one told me that Asterisk belonged on dedicated hardware before you, so I didn't know. - I'm just not very rich and try to integrate some things I need in my machine (don't worry, I did not framebuffered or X.orged it yet) because I cannot afford to buy another one (yes, even the 200€ one)... The part you don't want to know is how many people I had to kill in order to get my TDM400 card, until I found out that other cheaper solutions existed. :-) We're just trying to help -- but if you insist on running Asterisk in a VM, then you're on your own. And I thank you for that (the helping part), you've found the deep cause of all my zaptel problems (Xen), so please don't leave me alone! ;-) To be a bit more constructive, I'd like to ask you or anyone that dared to try using Asterisk on a non-dedicated hardware, specifically those that tried on a machine hosting VMs the following: - If there is no way running Asterisk with Xen, what type of 'hypervisor' should I use in order not to have problems? KVM?, KQemu?, VMWare? - What type of problems should I expect if I dare to do that? (of course, Asterisk will be realtime-niced to make it more important) Thanks and sorry again for the misunderstandings, François. ___ --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 ___ --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
Re: [asterisk-users] Get sip response code
17 maj 2007 kl. 02.57 skrev Robert Lister: On Wed, May 16, 2007 at 05:46:21PM -0500, Greg Oliver wrote: If I push the response code back to the handset (Cisco 7960) then it is even more unhelpful as it uses the same error message for all SIP error type response codes: Reorder but does not tell you why the call failed to set up. If it actually put the SIP response error on the display, that would be good, but it doesn't. (at least SIP 8.6 and prior software versions) In order to display the response on the handset, Cisco phones require that you have privileges and CTI control over the phones. The only un-authenticated things you can display through the phones are through the services and directories keys. Unfortunately, they are keeping that locked up since they want you to use them with their system. Hopefully they will change their minds one day. Yes. I know that... This is exactly the limitation I am trying to work around, by being able to send back meaningful tones to the phone from asterisk in-band rather than sending back the SIP response codes which all get displayed by the handset as Reorder which is completely useless in informing the user what's wrong. (And the US reorder tone sounds too much like the UK engaged tone anyway.) Even if the handset did display the SIP error response, I'm not expecting most users to understand the subtleties of what they all mean, so it seems better just to simplify it with a few well known tones most users are already familiar with (unobtainable, equipment busy, user busy, etc.) And it will behave in the same way regardless of the model of handset. (Call worked/Busy/Call failed...) Unfortunately Dial() DIALSTATUS is a bit limited in that if call setup fails for some reason, it mostly returns CONGESTION. Playing a congestion tone for perhaps 12 different call setup problems including misdials, doesn't help either. I want to play the right tone (for, say, unobtainable, equipment busy, etc.) The ISDN gateway I am using goes to great pains to send back the correct SIP response to asterisk, which then just reports it as CONGESTION which is a bit limiting. The SIP response code is displayed on asterisk's console, I just cannot see a way to get at it in the dial plan There are multiple ways to get the result of a dial() operation. The most detailed way is to read the HANGUPCAUSE channel variable, which is translated from each channels signalling protocol response code. As the ISDN gateway, Asterisk goes through great pains to translate SIP codes to the ISDN cause codes which we use as the esperanto code within the PBX core, which as you know is multiprotocol. The translation from SIP to the cause codes follows available RFCs and where those not cover the code, Cisco's documentation. I am sure that those are the same as your ISDN gateway adheres to. So in order to do something useful, use the HANGUPCAUSE channel variable. /O ___ --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
Re: [asterisk-users] Outside lines are just not happening...
Stephen Bosch wrote: And what do you do when they say: We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye. Ask for the supervisor. Don't quit pestering them until you get someone who can help you. DON'T QUIT. You'd be surprised what persistence will get you (that is, more than just disconnect supervision) My phone company is owned by two brothers, and it was one of them who told me that. I'm borked. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EUA voip provider
Hi some knows one good EUA voip provider with land lines? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone tested the new Sony Ericsson P1 phones..
Rosli Sukri wrote: Hi, Has anyone on this list tested out the new SE P1 phones (http://www.uncrate.com/men/gear/cell-phones/sony-ericsson-p1/). It says it supports VOIP, wonder if it is working with asterisk. Nice phone, wondering what it will cost when it gets released. So to answer your question, not very likely that someone has, since it's not available yet -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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
Re: [asterisk-users] OK to have Asterisk and clients behind firewalls?
On Thu, 17 May 2007, Vincent Delporte wrote: Hi To investigate the UNREACHABLE issue I'm having, I need to have confirmation that it's OK for the Asterisk server to be behind a NAT router, and also have clients elsewhere on the Net behind their own NAT router? Yes, it's OK... I know that clients must use STUN to resolve their public IP and punch UDP holes in their firewall, but is there something special that must be done in the configuration of Asterisk so it knows it's living in a private network, behind a NAT router? Yes. You need to do a few things. Firstly, you need the asterisk server on a static IP address on the inside, so make sure it doesn't get it's IP address from the local DHCP server. Next, you need to enable port-forwarding on your router. You need to forward port 5060 and 1 through 2 to the internal IP address of your asterisk box. Finally, you need to tell the asterisk box that it's on the inside of a NAT firewall. In sip.conf, you need 3 additional lines: nat=yes localnet=192.168.4.0/24 externip=1.2.3.4 You need to change localnet and externip to suit your network settings. It goes iwthout saying that you also need a static IP address on the internet connection that the asterisk server sits behind (but not for the phones) If using IAX then you just need to add port 4569 to the port forwarding rules on your firewall/router. And if someone knows of tools to investigate SIP issues, especially a text-based sniffer (no X available in the Asterisk live CD I'm using), I'm interested :-) tcpdump is the basic tool, but tetheral (now called wireshark, but I don't know what it's text-mode version is called - maybe the same) You can also capture packets with tcpdump to a file, then analyse them with a GUI enabled sniffer on a differnt workstation afterwards if required. PS: FWIW, extension 203 (softphone) and 204 (IP phone) are both located on the same network and behind a NAT router, and both connect out to an Asterisk server somewhere on the Net behing its own NAT router: slast*CLI sip show peers Name/username HostDyn Nat ACL Port Status 204/20482.237.x.y D 5060 UNREACHABLE 203/20382.237.x.y D N 46838OK (925 ms) I'd check the settings on the soft phone... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR is not written
There is a creation for Master.csv in /var/log/asterisk/cdr-csv and its filled with data ,but there is no pushing to mysql,asteriskcdrdb table cdr How or what is the procedure to let the data enters mysql . Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Thursday, May 17, 2007 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR is not written On Thu, 2007-05-17 at 10:54 -0700, Khaled Chehab wrote: I created Master.csv in /var/log/asterisk/cdr-csv ,even did not work,do freepbx make this problem,and how can I trouble shoot it. To get the CSV backend to pump CDRs into the /var/log/asterisk/cdr-csv/Master.csv file, you need to: a. in cdr.conf, in the general section, the enable=yes can optionally be there. just make sure it does ***NOT*** say enable=no b. in cdr.conf, the [csv] section must be there, and not commented out. I don't know what freepbx does to cdr.conf, but that file is a good place to start checking. BTW, doesn't freepbx have a mailing list? Wouldn't it be better asking there? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams Sent: Wednesday, May 16, 2007 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM]RE: [asterisk-users] CDR is not written I fought this for a bit when I found if the file Master.csv didn't exist, it wouldn't create it on it's own. I created an empty csv file, CDR started writing. Ken From: [EMAIL PROTECTED] on behalf of Khaled Chehab Sent: Thu 5/17/2007 10:50 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR is not written I installed asterisk 1.4.4 final ,but the cdr is not written any patch or tweaking can be done 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. * * 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. * ___ --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 * 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
Re: [asterisk-users] HPEC audio clipping
2007/5/16, Stephen Bosch [EMAIL PROTECTED]: As you mentioned, try implementing it only on certain channels, if you can. -Stephen- Hi, Unfortunately, I understood you couldn't allocate HPEC on per channel basis. But anyway, I agree we have to try something. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: zaptel huge irq problem
SD == Stephen Davies [EMAIL PROTECTED] writes: SD Hi, I want to quickly mention that I've had great success with SD running Asterisk in the under-appreciated Linux-VServer SD environment. I just want to do an AOL here: me too! Linux-vserver is great for asterisk, although we will probably migrate to OpenVZ. /Benny ___ --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
Re: [asterisk-users] G729 Transcoding problems
Anyone? some debugging help would be good otherwise I'm in the process of building a new asterisk box and hoping that getting away from Trixbox will help me CentOS 4.4 Asterisk 1.2.18 Zaptel 1.2.17 X100p (until I'm happy to swop the box into production then I will use my Sangoma A200) FreePBX 2.2.1 - I happen to like this GUI Thus far I've found that FreePBX seems much quicker than the Trixbox 2.0 build even with FreePBX upgraded to 2.2.1. Not got the the G729 install yet but I will use the free version. I've also found that the free G729 codec also fails in the same way as the Diguim one - so I conclude it is something wrong with my asterisk box. - Original Message - From: Wireless To: asterisk-users@lists.digium.com Sent: Wednesday, May 16, 2007 2:04 PM Subject: [asterisk-users] G729 Transcoding problems Hi I've bought the Digium g729 codec and have installed it correctly (I think) voip*CLI show g729 0/0 encoders/decoders of 3 licensed channels are currently in use if I do an echo test either from a sip Cisco 7960 or another hard phone (unbranded) using g729 it sometimes works and sometimes the announcement about the test (echo-test.gsm) fails part way though but the test continues to work ie I can hear the echo test - if Asterisk doesn't crash first! If I place a call from the Cisco phone or other phone using g729 / SIP into my * server and then out to my service provider using IAX2 / GSM asterisk restarts and the call fails I'm running a Trixbox 2.0 system but I have mannually patched it to Asterisk 1.2.18 Zaptel 1.2.17.1 Digium HPEC 8 (9 is not working right) Sangoma A200 with the wanpipe-3.1.0.p21-zaptel-patched drivers installed I've tried recompiling all these too. Since installing the g729 I've tried the different cpu types (the machine is a P3 650MHz) and found that the i686 works most reliably. any help much appreciated as I've search the Internet and this mailing list and am still stuck Harvey -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. -- ___ --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 ___ --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
Re: [asterisk-users] zaptel huge irq problem
Hi, I don't want multiple instances of Asterisk. My goal here is to make Asterisk and its Zaptel hardware run nice on a machine that is not dedicated and also hosts VM. I had lot's of problems with Xen, as the host runs a modified kernel that has apparently issues with the interrupt handling (at least). My question was more of what kind of hypervisor I should use for Asterisk/Zaptel not to have problems like it has in Xen. François. Tzafrir Cohen wrote: On Thu, May 17, 2007 at 07:26:10PM +0200, François Delawarde wrote: Hi, Why are you so determined to use Asterisk in a VM? You're asking for trouble. Asterisk belongs on dedicated hardware. I actually want to use Asterisk in a machine HOSTING a VM (that's what I implied with the Dom-0 thing I said earlier), sorry for the misunderstanding. I agree with you that given the state of advancement of just about any 'virtualizer', I would have to be totally stupid to try running Asterisk inside a VM. (I also wouldn't have asked here in the first place, as I would have been totally certain that problems came from the virtualizer itself) What kind of separation do you really need? Xen, VMWare and such are big cannons here. Every virtual machine will consume fixed ammount of memory. There is a considerable overhead for hardware access. It allows you things like running different OS/distribution on each guest. But for some reason I'm not sure you really need that? Will the users have direct acces to the dialplan and the rest of the configuration? If not: just run a single instance of Asterisk. If you do need multiple asterisk instances, verver or openvz might help you to give a separate container for that user's personal usage. Stephan has mentioned in this thread he set up several Asterisk-es on a vserver system. -- _ François Delawarde Ingeniero de red Tel: 918.03.92.51 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _ WIRELESS MUNDI http://www.wirelessmundi.com/ C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid 28760 TRES CANTOS (Madrid) Tlf./Fax: (+34) 918 03 92 51 La información contenida en este mensaje y en sus archivos adjuntos es CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda expresamente prohibida la utilización de la misma por cualquier persona distinta de los destinatarios de esta comunicación. Si usted ha recibido este mensaje por error le rogamos que lo comunique inmediatamente a WIRELESS MUNDI y lo borre al igual que todos sus documentos adjuntos. El correo electrónico no puede asegurar la confidencialidad ni la integridad de sus mensajes por lo que WIRELESS MUNDI no se hace responsable de tales errores u omisiones. --0-- All information in this message and its attachments is confidential and may be legally privileged. Only intended recipients are authorized to use it. If you have received this transmission in error, please notify WIRELESS MUNDI immediately and delete this message and its attachments. E-mail transmissions are not guaranteed to be secure or error free and WIRELESS MUNDI does not accept liability for such errors or omissions. ___ --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
Re: [asterisk-users] zaptel huge irq problem
I will try VMWare then. Does it support AMD-V or Intel VT virtualization at processor level for speedup? If so, I guess there is at some point some kernel code (module like KVM or special kernel like Xen) that could provoke some similar problems with zaptel interrupts. Could there be some issues here or am I totally wrong again? François. Stephen Bosch wrote: François Delawarde wrote: And I thank you for that (the helping part), you've found the deep cause of all my zaptel problems (Xen), so please don't leave me alone! ;-) To be a bit more constructive, I'd like to ask you or anyone that dared to try using Asterisk on a non-dedicated hardware, specifically those that tried on a machine hosting VMs the following: - If there is no way running Asterisk with Xen, what type of 'hypervisor' should I use in order not to have problems? KVM?, KQemu?, VMWare? The only one I would bother with is VMWare Server. It is solid, proven technology, and they have a big team of very talented engineers who have worked years to get the virtualization to the point where it can be sold as an enterprise grade product. If I were to try virtualizing anything, it would be on VMWare Server. - What type of problems should I expect if I dare to do that? (of course, Asterisk will be realtime-niced to make it more important) Well, in particular anything that expects unfettered access to hardware (as most realtime applications which rely on interface cards do) is going to be vulnerable to the proclivities of the hypervisor. Virtualization is still mostly rocket science. I have no doubt that it is the future and one day everything will run in virtualized environments -- but we're still a bit away from that. Virtualization makes financial sense when you have 20 database servers running at 10% utilization; you can drop your hardware requirements by at least a third... but for systems relying on dedicated hardware, I would be very careful (again -- I speak from ugly experience here). -Stephen- ___ --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 -- _ François Delawarde Ingeniero de red Tel: 918.03.92.51 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _ WIRELESS MUNDI http://www.wirelessmundi.com/ C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid 28760 TRES CANTOS (Madrid) Tlf./Fax: (+34) 918 03 92 51 La información contenida en este mensaje y en sus archivos adjuntos es CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda expresamente prohibida la utilización de la misma por cualquier persona distinta de los destinatarios de esta comunicación. Si usted ha recibido este mensaje por error le rogamos que lo comunique inmediatamente a WIRELESS MUNDI y lo borre al igual que todos sus documentos adjuntos. El correo electrónico no puede asegurar la confidencialidad ni la integridad de sus mensajes por lo que WIRELESS MUNDI no se hace responsable de tales errores u omisiones. --0-- All information in this message and its attachments is confidential and may be legally privileged. Only intended recipients are authorized to use it. If you have received this transmission in error, please notify WIRELESS MUNDI immediately and delete this message and its attachments. E-mail transmissions are not guaranteed to be secure or error free and WIRELESS MUNDI does not accept liability for such errors or omissions. ___ --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
Re: [asterisk-users] queue_exec: Unable to join queue
lenz wrote: I would try one of the two things: 1. adding a hint for the Local/[EMAIL PROTECTED] channels 2. using the = for queue members member = Agent/1001 member = Agent/1002 member = Agent/1003 Does this change anything? Hi Lenz thanks for your suggestions. I tried them both individually and together - no change. After a restart of Asterisk (now 1.4.4), I see the following on the first call with no callerid: app_queue.c:3541 queue_exec: Unable to join queue 'enidan' I then do a module reload app_queue, and everything is working fine. A show queue before reloading: enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet No Callers Does anyone know what (Invalid) means in this context? I'll check the code myself, but just in case someone recognises it. After a reload: enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet No Callers /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID not detected by TDM22B
Hi, I am using asterisk 1.4 with Digium TDM22B card. My system is running well except CALLERID. I have tried all options for cidsignalling, cidstart but no luck. Btw, I am living in Malaysia. From google web site, I found some one having the same problem with new version of asterisk but not in old versions. I do not want to try the old versions of asterisk. I really appreciate if someone can help me to solve the problem Regards ASLAY___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as SIP Provider
Can someone help me use Asterisk to provide SIP trunking to another PBX? I'd like to insert an Asterisk platform between a network PRI and an existing PBX. Incoming calls would (based on DNIS in a SQL table) go to PBX A on a PRI channel or PBX B as SIP. Outgoing calls from either PBX would select an available network PRI channel. --Don Don Kelly CT Magic 612 843-6060 888 Don Kell(y) 612 843-6061 fax ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk vs. Shoretel
Hi, Someone who has had experience with shoretel VoIP systems, can you please give me a run down of how Asterisk is either better or worse? I am completely unfamiliar with Shoretel systems, but someone had suggested we look into them. I said, you bring your Shoretel features, and I'll show you 10 things Asterisk does or can do that Shoretel doesn't do. I still believe that's possiblehowever, I thought I'd shoot an e-mail out here to see what experience others have. ___ --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
RE : [asterisk-users] CallerID not detected by TDM22B
Hello Aslay, In some country, this feature is a paid option from the TELCO side. In France the analog lines have not this feature enabled in standard, only the digital lines . Are you sure that it's actualy available in your case ? Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de aslay-pinwee Envoyé : vendredi 18 mai 2007 14:23 À : asterisk-users@lists.digium.com Objet : [asterisk-users] CallerID not detected by TDM22B Hi, I am using asterisk 1.4 with Digium TDM22B card. My system is running well except CALLERID. I have tried all options for cidsignalling, cidstart but no luck. Btw, I am living in Malaysia. From google web site, I found some one having the same problem with new version of asterisk but not in old versions. I do not want to try the old versions of asterisk. I really appreciate if someone can help me to solve the problem Regards ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call-limit=2 , call counter not reset to zero after hangup
Hi all,
There is a case in which the call counter is not set to zero for a sip peer
(incoming call). Here is the scenario.
Dialplan:
exten= 1,1,Dial(SIP/U1)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10,1)
exten= 1,3,Hangup
exten= 10,1,Voicemail()
If a user just registered with my asterisk and due to some reason after
sometime the user's ATA gets turned off. But asterisk does not know that
becoz there is some time left for the ATA to re-register itself. Now if
anyother user calls U1 he will get a dialstatus of NOANSWER and will be
thrown to voicemail. after voicemail has ended and the calling user has hung
up. the incoming call limit counter for the called user (in this case U1) is
not reset to zero. This only happens in this particular case. otherwise the
call counter for peer and user works fine.
So if anybody knows how to fix this prob, plz share the solution. Can i set
the call counter to zero from the dialplan? or is there anyway to know that
the user is registered or not before dialing that user?
I am using asterisk 1.4.2
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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[asterisk-users] Outside lines--*solved*!
After spending a couple of hours on the phone with Digium support, monkeying with battery threshold and several other settings, we figured out that the line was too noisy. I worked with our local telco to resolve that, and it's fixed! Thanks for all the suggestions; I'm learning *so* much about this system! J. David Bavousett System Administrator Abilene Library Consortium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. Apparently this is a router related issue. Its a D-Link DI-624 router. This happens even when there is no Internet activity at all. How can this issue be resolved. What are the reasons for losing contact with the router. Is there some interference at port 5060, bad wiring, or something else? -- Zeeshan A Zakaria ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Does it loose it's IP address consistently and at a designated interval. I've seen something similar in a number of various cases where it was always an issue of the 'client' device blocking the DHCP renew traffic. But when it went into rebinding it would drop $filter and allow the dhcp traffic back through. I would say check to see if it occurs at a regular interval and compare that with the lease times. Also consider it's sometimes a good idea if the phone supports it (I don't know if your does) but setting up a cheap syslog host that can catch the syslog messages from the unit. Some units will log why they dropped their network (ie dhcp) or something. This was just a wild stab. On May 18, 2007, at 8:51 AM, Zeeshan Zakaria wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. Apparently this is a router related issue. Its a D-Link DI-624 router. This happens even when there is no Internet activity at all. How can this issue be resolved. What are the reasons for losing contact with the router. Is there some interference at port 5060, bad wiring, or something else? -- Zeeshan A Zakaria ___ --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 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --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
RE: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. We have also had this happen at client sites. I don't think it's a router issue, since we've had the same thing occur using a variety of different routers. There are 2 possible reasons for it: 1) The firmware version on the GXP2000 isn't doing DHCP queries properly - when it connects to the server to renew its IP, instead it thinks the server's given it a different one (which of course it hasn't). I'm guessing the phone drops the network interface then brings it back up. 2) The auto check firmware every option was known to cause the phone to reboot in some firmware versions when the phone checked for new firmware/config from the TFTP server. We worked around the issue at one place by giving all the phones static IPs (and configuring them in the phones manually) and disabling auto check firmware. At another location where static IPs would have been impractical (some staff take them home at weekends), we simply upped the lease time on the DHCP server to 7 days. Of course, you may find that simply trying different versions of the firmware resolve those issues. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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
Re: [asterisk-users] Outside lines are just not happening...
Brian Capouch wrote: Stephen Bosch wrote: And what do you do when they say: We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye. Ask for the supervisor. Don't quit pestering them until you get someone who can help you. DON'T QUIT. You'd be surprised what persistence will get you (that is, more than just disconnect supervision) My phone company is owned by two brothers, and it was one of them who told me that. I'm borked. Well, your situation is special (and I was thinking of your situation when I said: most large telcos have a programming department ;) -Stephen- ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
As others have said, it does sound like a DHCP issue.. you can try increasing the lease time, or giving it a static IP address. On 5/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. Apparently this is a router related issue. Its a D-Link DI-624 router. This happens even when there is no Internet activity at all. How can this issue be resolved. What are the reasons for losing contact with the router. Is there some interference at port 5060, bad wiring, or something else? -- Zeeshan A Zakaria ___ --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 ___ --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
RE: [asterisk-users] Asterisk vs. Shoretel
I'm kind of curious on this one myself. Any direct comparisons online? I wasn't really paying much attention to them until they announced the Salesforce.com tie-up. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Friday, 18 May 2007 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk vs. Shoretel Hi, Someone who has had experience with shoretel VoIP systems, can you please give me a run down of how Asterisk is either better or worse? I am completely unfamiliar with Shoretel systems, but someone had suggested we look into them. I said, you bring your Shoretel features, and I'll show you 10 things Asterisk does or can do that Shoretel doesn't do. I still believe that's possiblehowever, I thought I'd shoot an e-mail out here to see what experience others have. image001.gif___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Zeeshan Zakaria wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and The phone is probably renewing it lease with the DHCP. Set your expire time to a larger number. I've got mine set for 30 days. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Quoting Doug Lytle [EMAIL PROTECTED]: Zeeshan Zakaria wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and The phone is probably renewing it lease with the DHCP. Set your expire time to a larger number. I've got mine set for 30 days. That is probably the cause but it also sounds like a flawed implementation - the ip should not just go away while its being renewed. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
On 5/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is effecting his business. Sounds like DHCP to me. I've not had this problem with a GXP-2000. If it were me, I would try setting the IP address to a static IP to rule out any kind of DHCP weirdness. Now, if it still happens, then it's not really losing its IP; instead, it will be losing the connection. Or it could be unregistering with the Asterisk server for some reason, and then re-register 30-60 seconds later. That's still bad, but it's different than losing one's IP. One way you may be able to more accurately diagnose the problem would be to run Ethereal or some other packet sniffer and see if the voice packets get through your router. If not, fix or replace your router. If so, you'll need to do more detective work to see if you have a phone problem, configuration problem, cabling problem, bad port on the switch, etc. Hope that helps, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN: long delay when making outbound calls
Hi, I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet card (with ports in PTP mode). I noticed a long delay when making outbound calls, more precisely between (taken from Asterisk CLI) Called 1/X/s and mISDN/1-u43 is proceeding passing it to SIP/8-5486 I searched on misdn.org but found nothing. I'd like to understand if this delay is caused by telco line or the driver and in the latter case if there is a way to shorten this delay. TIA Giorgio Incantalupo -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 82
If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 17 May 2007 11:22:17 -0400 From: Race Vanderdecken [EMAIL PROTECTED] Subject: RE: [asterisk-users] cpu usage for G.729 codec To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii G.729 is a compromise of bandwidth vs. CPU power. It takes more CPU but less bandwidth. It depends on what your want to do with the G.729. Pass through does not involve any transcoding, that I know of, so it is just an RTP packet movement, no different than the cost of other pass through codecs. I did work on converting G.729 to G.711 to disk storage in real time and that took about 3% of a Xeon CPU for full duplex. Memory wise each convert call might have used 640KB in buffers and trash, but not much. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] web app to playback recorded phone calls.
1 of our customers records all phone calls and needs to be able to be played back via a searchable web app. I tried ARI but it is very limited. Anyone have any ideas? ___ --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
Re: [asterisk-users] cpu usage for G.729 codec
(Note: resending with proper Subject) If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 17 May 2007 11:22:17 -0400 From: Race Vanderdecken [EMAIL PROTECTED] Subject: RE: [asterisk-users] cpu usage for G.729 codec To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii G.729 is a compromise of bandwidth vs. CPU power. It takes more CPU but less bandwidth. It depends on what your want to do with the G.729. Pass through does not involve any transcoding, that I know of, so it is just an RTP packet movement, no different than the cost of other pass through codecs. I did work on converting G.729 to G.711 to disk storage in real time and that took about 3% of a Xeon CPU for full duplex. Memory wise each convert call might have used 640KB in buffers and trash, but not much. -- (C) Matthew Rubenstein ___ --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
Re: [asterisk-users] HPEC audio clipping
Olivier wrote: 2007/5/16, Stephen Bosch [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: As you mentioned, try implementing it only on certain channels, if you can. -Stephen- Hi, Unfortunately, I understood you couldn't allocate HPEC on per channel basis. But anyway, I agree we have to try something. Kevin just announced the release of a patched version that is supposed to correct the problem. Kevin P. Fleming wrote: I have just placed HPEC version 9.00.003 onto the Digium FTP site; this release cures the recently discovered bug where incoming audio was choppy or lost completely under specific circumstances (did not affect all users, but if a user's system had the problem it would have it on all calls). -Stephen- ___ --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
Re: [asterisk-users] CallerID not detected by TDM22B
MessageHello, My TELCO already enable the service. I also verify by connectiong the PSTN line direct to a phone with LCD display. I can see the caller number displayed on the LCD screen. I believe that the root problem is the setting on the zapata.conf (cidsignalling, cidstart), which I am not sure what to set for my country TELCO. ASLAY - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, May 18, 2007 8:47 PM Subject: RE : [asterisk-users] CallerID not detected by TDM22B Hello Aslay, In some country, this feature is a paid option from the TELCO side. In France the analog lines have not this feature enabled in standard, only the digital lines . Are you sure that it's actualy available in your case ? Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de aslay-pinwee Envoyé : vendredi 18 mai 2007 14:23 À : asterisk-users@lists.digium.com Objet : [asterisk-users] CallerID not detected by TDM22B Hi, I am using asterisk 1.4 with Digium TDM22B card. My system is running well except CALLERID. I have tried all options for cidsignalling, cidstart but no luck. Btw, I am living in Malaysia. From google web site, I found some one having the same problem with new version of asterisk but not in old versions. I do not want to try the old versions of asterisk. I really appreciate if someone can help me to solve the problem Regards ASLAY -- ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE212P octastic initialization failure
Hi, I'm trying to get a TE212 working on a Dell PowerEdge 1850 running Debian etch using the latest release of libpri (1.4.0), zaptel (1.4.2.1) and asterisk (1.4.4). The initilization of the Octasic echo canceller seems to fail when the wct4xxp module is loaded. [...] VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize [...] By looking in the zaptel code, this error value (0x00103017) means cOCT6100_ERR_OPEN_EXTERNAL_MEM_BIST_FAILED. Is anyone familiar with that problem ? Thanks for your help. --- TE212P card: jumpers are set to E1 mode and nothing is connected to that card at the moment. # uname -a Linux ditti-voipa-serv-1 2.6.18-4-amd64 #1 SMP Fri May 4 00:37:33 UTC 2007 x86_64 GNU/Linux # cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 # cat /proc/interrupts CPU0 CPU1 0: 42385 0IO-APIC-edge timer 6: 3 0IO-APIC-edge floppy 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 64 0IO-APIC-edge ide0 169: 0 0 IO-APIC-level uhci_hcd:usb1 177: 0 0 IO-APIC-level uhci_hcd:usb2 185: 0 0 IO-APIC-level uhci_hcd:usb3 193: 19 0 IO-APIC-level ehci_hcd:usb4 201: 2148 0 IO-APIC-level ioc0 217: 1153 0 IO-APIC-level eth1 225: 160247 0 IO-APIC-level wct2xxp NMI: 64 42 LOC: 42340 42317 ERR: 0 MIS: 0 # dmesg [...] Found TE2XXP at base address fe7ffc00, remapped to c2004c00 TE2XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x7daa5400 Reg 1: 0x7daa5000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE210P (3rd Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present Failed to get chip capacity, code 0010305e! Unsupported channel capacity found on VPM module (0). Completed startup! [...] # ztcfg -v Zaptel Version: 1.4.2.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 62 channels configured. # ___ --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
Re: [asterisk-users] how to define a key to decline incoming call
Anselm Martin Hoffmeister wrote: Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]: Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a SIP/2.0 486 Busy Here message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the call. Acting on feature.conf we succed in defining keys for blind transfer or attended transfer: the last thing we need is the ability to drop an incoming call without answering it. Is there any way to define a key (or double-key, i.e. *4) to send back a SIP/2.0 486 Busy Here message ? thanks in advance, Please have a look at http://www.voip-info.org/wiki/view/Asterisk+Cmd+Dial Especially the M() parameter. There is an example 2 that needs only little changing to match your idea. You could, for example, wait 2 secs in that macro before bridging the calls - and reject call if 1 is pressed within 2 seconds, or similar. Of course this prepends 2 seconds to any call bridging. For the Siemens Gigasets, most of them (and I do not know the C450IP, only the C450) show a soft button Ignore which meens stop ringing, but do not tell the caller. Combined with a Dial() timeout and following voicemail this works like stop disturbing, eventually the voicemail will take the call. No idea about the ATCOM though, so having that Macro stuff might be the most universal method available. Do these sets not have a Do not disturb button? For sets that do, what is the behaviour when one presses the DND button while the phone is ringing? -Stephen- ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Thanks for all the replies. I've updated client's router firmware, which was very old. Now I've put MoH on for the whole day and see if it happens again. MoH was doing the same thing, i.e. going down and coming back up. When it went down, on the screen you could see message saying 'No IP', and when it came back up, same old IP was back. Phone's firmware is the latest one, which is on their website, i.e. 1.1.1.14. I know there exists some latest but still beta version of a firmware, but I'll stick with this one. There is no pattern when it goes down, it may does it twice an hour or not do it for many hours. But I've noticed that Automatic Upgrade Option was set to 60 min, I've now set it for 1440 minutes. Does automatic upgrade means only firmware upgrade or config upgrade as well. Can I have config upgrade sooner than 1440 minutes on this phone? ___ --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
Re: [asterisk-users] web app to playback recorded phone calls.
Hi Eric, We used a quick hack to Asterisk Queue/CDR Log Analyzer (http://www.micpc.com/qloganalyzer/) to turn the Uniqueid field into a hyperlink to the .WAV file for the call. We only use this internally for our call centre, it is not customer ready as all call details are revealed and not all hyperlinked Uniqueid's lead to a recording. regards, Drew Hall, Eric M. wrote: 1 of our customers records all phone calls and needs to be able to be played back via a searchable web app. I tried ARI but it is very limited. Anyone have any ideas? ___ --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 -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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
Re: [asterisk-users] Maximum retries exceeded on transmission
13 apr 2007 kl. 16.45 skrev Brian Jones: I've encountered a similar problem with Cisco equipment. The Cisco proxy was not replying to Asterisk with an ACK after * sent an OK. Since version 1.2.14, * was changed so that not receiving an ACK to an OK is considered a FATAL error. The specific change that causes this problem is in sip_answer() in chan_sip.c: res = transmit_response_with_sdp(p, 200 OK, p-initreq, 2); Changing the 2 to a 1 will probably fix it. Note that this is NOT a bug in * but improper implementations--either caused by latency, or a software bug (not sending an ACK). Perhaps it might be beneficial to have an option in sip.conf to change how * handles not receiving an ACK? I know... it's someone else's problem, but might help those of us stuck with buggy implementations in production environments. :) Answering late, but still answering to this mail from april that was highlighted to me by an Asterisk user. This change is *not recommended* and will in worst case cause Asterisk to have channels hanging with open UDP ports and eventually break your system. Not sending an ACK on an INVITE-200 OK- ACK transaction is and should be a fatal error. If you change this, you really need to know what you're up to. (And don't request help on the bug tracker :-) ) /Olle ___ --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
Re: [asterisk-users] how to define a key to decline incoming call
On 5/18/07, Stephen Bosch [EMAIL PROTECTED] wrote: Do these sets not have a Do not disturb button? For sets that do, what is the behaviour when one presses the DND button I have some Aastra's (a mixture of 480i's, 9112i's, and 9133i's) and whenever I program a key as do-not-disturb, I can press the key while my phone is ringing and it immediately passes the call to the next step in my dialplan (Voicemail/busy). It also does this if I press the Hangup key on the Aastra. -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query about DTMF generate
Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. ___ --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
Re: [asterisk-users] TE212P octastic initialization failure
On May 18, 2007, at 9:53 AM, Francois Deppierraz wrote: Hi, I'm trying to get a TE212 working on a Dell PowerEdge 1850 running Debian etch using the latest release of libpri (1.4.0), zaptel (1.4.2.1) and asterisk (1.4.4). The initilization of the Octasic echo canceller seems to fail when the wct4xxp module is loaded. [...] VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize [...] By looking in the zaptel code, this error value (0x00103017) means cOCT6100_ERR_OPEN_EXTERNAL_MEM_BIST_FAILED. Is anyone familiar with that problem ? No. It looks like something you should definitely talk to Digium Support about that though. They should be able to help you find what is wrong. Matthew Fredrickson ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Physical connection is fine because customer's computer is also connected through the phone's inline ethernet port. When phone was losing IP, computer was still working fine on the Internet. But after the changes I made earlier today, as mentioned previously, it seems to be working fine so far. I'll check it again from them at the end of the day. ___ --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
Re: [asterisk-users] Phone losing IP address for a few seconds but doesn't drop call
Hi, Sounds like a flaky physical connection to me. Try new patch cables (DO NOT EVER use homemade patch cables, they cost too much) and a different jack if in an office environment. If that doesn't fix it, try locking the switch port and/or the phone port speed/duplex settings and turn off auto-negotiation. Try a different make/model of switch (insert between phone and router). regards, Drew Zeeshan Zakaria wrote: Thanks for all the replies. I've updated client's router firmware, which was very old. Now I've put MoH on for the whole day and see if it happens again. MoH was doing the same thing, i.e. going down and coming back up. When it went down, on the screen you could see message saying 'No IP', and when it came back up, same old IP was back. Phone's firmware is the latest one, which is on their website, i.e. 1.1.1.14 http://1.1.1.14. I know there exists some latest but still beta version of a firmware, but I'll stick with this one. There is no pattern when it goes down, it may does it twice an hour or not do it for many hours. But I've noticed that Automatic Upgrade Option was set to 60 min, I've now set it for 1440 minutes. Does automatic upgrade means only firmware upgrade or config upgrade as well. Can I have config upgrade sooner than 1440 minutes on this phone? ___ --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 -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --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
Re: [asterisk-users] how to define a key to decline incoming call
You've got 2 possibilities : 1. If you want something which is channel or phone type independant and works for analog phones, for instance, you've got to use AMI to tell Asterisk to hangup this specific call. 2. If you accept something which depends on phone type, you've got to look in phone arcanes and look for Reject or DNS menus or try blind transfer for a specially created extension ;-) My 2 cents ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Asterisk console loop?
I believe this is a Asterisk console bug, but thought I would run it through here first. I can get Asterisk into a tight loop 100% of the time. Here is what I know... First I have verbosity set to 20 in the asterisk.conf file. I believe that has some bearing on this issue for two reasons. One it doesn't occur if I turn verbosity off and two, there is no updates to the console occurring without verbosity. Next, I connect to my Astlinux box using putty on a Windows XP machine (yes, windows is involved in this but it may happen with other clients, just not tested) and fire up asterisk console with "asterisk -r" Next, I put my Windows machine into suspend without shutting down putty. This leaves the connection in an open state on the Linux box. Next, I have a call generated on my Digium TDM401 card via zap. This may also work with SIP but it is not as consistent. At this point, the box stops responding. If I call the Zap Channel line, I hear the SIT tones and the message starting "This system..." in stutter format, like the machine is trying to push it out but the box is at 100%. The console is not responsive and I cannot troubleshoot it any further. There is no dump file created or any logging done on the box. I have no way to verify which asterisk process is exactly to blame. Here is my theory, The Zap Channel is trying to write via the console to a port which was not closed properly, therefore, it goes into a loop trying to contact that device. Like I said, this happens 100% of the time, I just cannot troubleshoot it further. If I close out the putty session properly, I do not have an issue. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zap fallback
I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Query about DTMF generate
gaurang sheladiya wrote: Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. Asterisk cmd SendDTMF: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SendDTMF General Google Search of www.voip-info.org: http://www.google.com/custom?tk=b44a2457968317bad5a3domains=www.voip-info.org Have a great day. -- Warm Regards, Lee ___ --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
Re: [asterisk-users] Query about DTMF generate
gaurang sheladiya wrote: Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. After reading your post again, I thought maybe this would be of use more than other links I provided: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+indications.conf -- Warm Regards, Lee ___ --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
Re: [asterisk-users] zap fallback
On May 18, 2007, at 11:50 AM, Steve Kennedy wrote: I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Asterisk does not do dialtone detection before it starts using a zap channel. That's probably why you are seeing this behavior. Matthew Fredrickson ___ --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
Re: [asterisk-users] zap fallback
On Fri, May 18, 2007 at 12:54:19PM -0500, Matthew Fredrickson wrote: On May 18, 2007, at 11:50 AM, Steve Kennedy wrote: I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Asterisk does not do dialtone detection before it starts using a zap channel. That's probably why you are seeing this behavior. Oh well, have to think of another way around this. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 4-port ATA
Can anyone recommend a 4-port ATA (preferably IAX2, but SIP would be okay). TIA for any recommendations/experience. ___ --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
Re: [asterisk-users] zap fallback
Steve Kennedy wrote: I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. The TDM400P does not detect dialtone. It also does not detect if there is voltage on the line. Make sure your lines are working. ___ --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
RE: [asterisk-users] SIP Hardware Phone
Lol - I think everyone starts on budgetones then moves to ciscos or polycoms etc. BTW I have 2 x 3line IP500's that I'm about to ebay this weekend if anyone in NY or near Manhattan wants to buy them. Just upgraded to the 6 line versions. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Wednesday, 16 May 2007 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Hardware Phone [EMAIL PROTECTED] wrote: Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY Polycoms seem great to me and according to much feedback I've read on this list. I had a budgetone 100 when I first started playing with Asterisk, but once I was sure I wanted to use it/learn it, I got a polycom. -- Warm Regards, Lee ___ --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 ___ --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
RE: [asterisk-users] CDR is not written
I fought this for a bit when I found if the file Master.csv didn't exist, it wouldn't create it on it's own. I created an empty csv file, CDR started writing. Ken From: [EMAIL PROTECTED] on behalf of Khaled Chehab Sent: Thu 5/17/2007 10:50 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR is not written I installed asterisk 1.4.4 final ,but the cdr is not written any patch or tweaking can be done 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. * winmail.dat___ --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
Re: [asterisk-users] zap fallback
i would force a timer on it.. dial (blah,30) maybe that would bypass , maybe not.. i actually think it wont.. another example of this problem is DNS echo '1.2.3.4your.favorite.itsp' /etc/hosts then Dial(SIP/[EMAIL PROTECTED]) DNS failing will BLOCK the call indefinitely... On 5/18/07, Steve Kennedy [EMAIL PROTECTED] wrote: On Fri, May 18, 2007 at 12:54:19PM -0500, Matthew Fredrickson wrote: On May 18, 2007, at 11:50 AM, Steve Kennedy wrote: I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Asterisk does not do dialtone detection before it starts using a zap channel. That's probably why you are seeing this behavior. Oh well, have to think of another way around this. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --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
RE: [asterisk-users] asterisk manager interface stability
Do you have any specific experience with astmanproxy? Can anyone give me an idea on number of simultaneous connections this can legitimately handle with ease? This has been around for a while by the looks of it but I haven't heard about it before. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Smith Sent: Wednesday, 16 May 2007 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] asterisk manager interface stability You guys all sound like you're talking about AstManProxy. See: http://www.voip-info.org/tiki-index.php?page=AstManProxy I'm not saying it is the solution to your problem per se, but I can't help but think of it when I read the descriptions of what people want (you even use the word proxy!). Figured I'd send this out in case someone hadn't seen it. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Subject: Re: [asterisk-users] asterisk manager interface stability The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 sniffer and player
Hi all, I was wondering if there is any IAX2 sniffer and decoder. Wireshark can decode and play RTP streams using G.711, and Cain Abel decodes and plays any kind of RTP stream. But I didn't find anyone that can decode IAX2 streams. Any programs?? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --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
Re: [asterisk-users] asterisk manager interface stability
Matt Florell wrote: On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. I've been considering writing something like this for a project that I'm thinking about doing that would require potentially high number of concurrent clients to consume AMI services. From your experience, does the software that you wrote require significant CPU to cache and then doll out the kind of volume of messages that AMI can send? One of the great parts about removing the broadcasting of AMI events outside of the Asterisk process is that the broadcast server process can exist on a separate physical server removing any kind of overhead on the Asterisk server. In my experience doing the proxy on the same machine uses less CPU resources than the same number of AMI connected clients, and doesn't have any of the deadlock issues that can happen with a lot of direct AMI connections. For my application(ACQS) I use MySQL as a storage engine for all of the recent events received and sent so that they can be independantly queried by any client apps that need to see them. MATT--- Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? -- Warm Regards, Lee ___ --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
Re: [asterisk-users] asterisk manager interface stability
On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote: Matt Florell wrote: On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. I've been considering writing something like this for a project that I'm thinking about doing that would require potentially high number of concurrent clients to consume AMI services. From your experience, does the software that you wrote require significant CPU to cache and then doll out the kind of volume of messages that AMI can send? One of the great parts about removing the broadcasting of AMI events outside of the Asterisk process is that the broadcast server process can exist on a separate physical server removing any kind of overhead on the Asterisk server. In my experience doing the proxy on the same machine uses less CPU resources than the same number of AMI connected clients, and doesn't have any of the deadlock issues that can happen with a lot of direct AMI connections. For my application(ACQS) I use MySQL as a storage engine for all of the recent events received and sent so that they can be independantly queried by any client apps that need to see them. MATT--- Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- ___ --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 ___ --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
RE: [asterisk-users] asterisk manager interface stability
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, 18 May 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk manager interface stability Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- ___ Wouldn't that make it highly inefficient? Is there no two way dialog or am I missing something? Basically if I have 100 end user clients that needed real time interaction with astproxy are you saying that each client would need to poll once per second (eg 100 polls per second) in order to see if something happened that second that was relevant to them?) Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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
Re: [asterisk-users] Re: DUNDi configuration problem
Thank you for the quick response. Do I need to create a route to the
other machine? like a trunk?
greetz, Tim
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial
Your mappings are wrong, this is for IAX, for SIP to work, it should be:
priv = dundi-priv-canonical,0,SIP,${NUMBER}@the real IP Address,nopartial
The rest looked ok I think.
Good luck.
JR
--
JR Richardson
Engineering for the Masses
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[asterisk-users] Who picked up with *8?
Is there a way to know who picked up a call using *8? A customer wants to know if someone is picking up their calls when they are not at their desk. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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
Re: [asterisk-users] Who picked up with *8?
Use the cdr's, who wont know who but at least which phone did it. Carlos Chavez wrote: Is there a way to know who picked up a call using *8? A customer wants to know if someone is picking up their calls when they are not at their desk. ___ --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 ___ --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
Re: [asterisk-users] asterisk manager interface stability
On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, 18 May 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk manager interface stability Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- ___ Wouldn't that make it highly inefficient? Is there no two way dialog or am I missing something? It is inefficient, but it is non-blocking which the AMI is not. having a separate listen and separate send processes removes the problems with AMI locking up on high volume Asterisk systems. Basically if I have 100 end user clients that needed real time interaction with astproxy are you saying that each client would need to poll once per second (eg 100 polls per second) in order to see if something happened that second that was relevant to them?) Not a problem for MySQL, that's what it does best. The astguiclient application can do 20+ queries per second per client depending on the application. I currently have one company with over 200 client applications(AJAX) sending 3000-4000 queries per second to the MySQL server, and it handles it just fine. On the client side, the load is not very high either, even on a PIII 700MHz machine. MATT--- Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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 ___ --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
RE: [asterisk-users] asterisk manager interface stability
Sweet thanks Matt. If there are any developers in Manhattan (or very nearby) who have experience with Astproxy and are looking for sweat equity ownership in a new Asterisk application get in touch. Also looking for someone with ROR UI skills but I might already have that role filled. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, 18 May 2007 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk manager interface stability On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, 18 May 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk manager interface stability Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- ___ Wouldn't that make it highly inefficient? Is there no two way dialog or am I missing something? It is inefficient, but it is non-blocking which the AMI is not. having a separate listen and separate send processes removes the problems with AMI locking up on high volume Asterisk systems. Basically if I have 100 end user clients that needed real time interaction with astproxy are you saying that each client would need to poll once per second (eg 100 polls per second) in order to see if something happened that second that was relevant to them?) Not a problem for MySQL, that's what it does best. The astguiclient application can do 20+ queries per second per client depending on the application. I currently have one company with over 200 client applications(AJAX) sending 3000-4000 queries per second to the MySQL server, and it handles it just fine. On the client side, the load is not very high either, even on a PIII 700MHz machine. MATT--- Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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 ___ --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 image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unsubscribe
Cristian López F. Integración y Tecnología - Terra Chile Phone: (56 2) 330 6966 movil: 56-92401759 E-mail: [EMAIL PROTECTED] Este correo y su contenido solamente interesan a las personas autorizadas de TERRA NETWORKS CHILE. Si usted fue receptor de este correo por error, por favor no lo tome en cuenta y avise al remitente. This message is solely of the interest of TERRA NETWORKS CHILE or its businesses. If you have received this e-mail by error, please ignore it and notify the sender.___ --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
RE: [asterisk-users] unsubscribe
Disclaimer at the bottom still looks ridiculous even in Spanish... LOL Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ Helping students on a mission. Graduation and beyond. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, May 18, 2007 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] unsubscribe Cristian López F. Integración y Tecnología - Terra Chile Phone: (56 2) 330 6966 movil: 56-92401759 E-mail: [EMAIL PROTECTED] Este correo y su contenido solamente interesan a las personas autorizadas de TERRA NETWORKS CHILE. Si usted fue receptor de este correo por error, por favor no lo tome en cuenta y avise al remitente. This message is solely of the interest of TERRA NETWORKS CHILE or its businesses. If you have received this e-mail by error, please ignore it and notify the sender. image001.jpg___ --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
Re: [asterisk-users] Who picked up with *8?
On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote: Use the cdr's, who wont know who but at least which phone did it. I tried following the CDR but if I dial extension 4000 and extension 4002 picks up the call using *8 the CDR says that extension 4000 ANSWERED the call. It does not say that 4002 did anything. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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
Re: [asterisk-users] asterisk manager interface stability
Matt Florell wrote: On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote: Matt Florell wrote: On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Matt Florell wrote: The issue has more to do with the sheer amount of data passed to the client from within the Asterisk application when you have 50-100+ clients connected to the AMI on full output mode. Running a system with FreePBX/Trixbox especially generates vast amounts of output that has to be generated on every AMI connection for every client. This is not trivial and can result in lockups very easily, although this has gotten much better since the early 1.0 versions. The new Asterisk Manager web API in 1.4 is a good step where sending of Actions does not require an actual Telnet conneciton to the AMI, but I think to be able to handle larger numbers of concurrent connections that a separate send-only and a separate receive-only type of interface be built where Asterisk would just output all AMI data to a single server-like application that would then broadcast it to all connected clients. This would remove the burden of so many connections going directly into Asterisk and would allow for much larger scaling of AMI-type applications that require real-time output of AMI events. I definitely agree here personally. Clients could connect to this proxy and subscribe to only the events that are interesting or applicable. As for how to go about doing this, I can't help you there. I did build a very specialized version of something like this 4 years ago for the astGUIclient project called the Asterisk Central Queue System(ACQS) It is based on 1.0 Asterisk but it still works with 1.2 and 1.4. It is limited in what it does, but it does scale much better than using direct AMI connections. I've been considering writing something like this for a project that I'm thinking about doing that would require potentially high number of concurrent clients to consume AMI services. From your experience, does the software that you wrote require significant CPU to cache and then doll out the kind of volume of messages that AMI can send? One of the great parts about removing the broadcasting of AMI events outside of the Asterisk process is that the broadcast server process can exist on a separate physical server removing any kind of overhead on the Asterisk server. In my experience doing the proxy on the same machine uses less CPU resources than the same number of AMI connected clients, and doesn't have any of the deadlock issues that can happen with a lot of direct AMI connections. For my application(ACQS) I use MySQL as a storage engine for all of the recent events received and sent so that they can be independantly queried by any client apps that need to see them. MATT--- Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- Cool. I hadn't thought of doing it that way. My idea was to somehow keep an in memory cache for client connections. As events were received from the AMI, poll a hash table in memory and forward the event to client connections who have registered interest in that event. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] xten will not send tones to * and i from sip phone
hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some sort of configuration out there to tell the xten softphone to work as expected? thanks! Then another problem! I used the i extension, plus _X and _X. to make sure I catch everything that is not propperly dialed. If I take the regular phones that are connected through the sipura ata, then dial 'exten = 700,1,Goto(default,s,1)' so that I get the asking for an extension to reach, I dial a wrong number and walla, its caight by one of my magic numbers! BUT, if I pickup the same phone, and just dial the same wrong number? I just get a busy signal! and there is nothing registered at the CLI even though I added DEBIG to the configuration! :s What can I do to make sure I always send an error sound and never again a busy signal? thanks! Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --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
Re: [asterisk-users] Dealing with 2 SIP providers
On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote: Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available What would be really cool, but require special code in the chan_sip dialer, would be automatic support of multiple providers in a similar fashion to the way Asterisk can ring two channels and only talk to the first to answer. You can't just do this with outgoing providers, because if you try to ring two at once, you may very well have the second one go to a voicemail and thus answer right away (because the first is ringing) and you would treat that as the success. What I have in mind is something like this: a) Invite to main provider b) Await some intermediate response, such as a RINGING code or some early media c) If you don't get that after a short timeout (more like 5 seconds) then INVITE the second provider d) Upon the receipt of a ringing or early media code from either, CANCEL the other. Now you would have to get your timings right because there could still be risk of doing something bad, such as a 2nd call going to voice mail or residual ringing making a call waiting on the recipient. (I don't know what typical 5ess do with a 2nd call that comes in while still ringing, anybody known?) Anyway, this could be a good course when a provider has known unreliability. Long timeouts and restarts are very annoying to 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
Re: [asterisk-users] asterisk manager interface stability
Matt Florell wrote: On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, 18 May 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk manager interface stability Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- ___ Wouldn't that make it highly inefficient? Is there no two way dialog or am I missing something? It is inefficient, but it is non-blocking which the AMI is not. having a separate listen and separate send processes removes the problems with AMI locking up on high volume Asterisk systems. Basically if I have 100 end user clients that needed real time interaction with astproxy are you saying that each client would need to poll once per second (eg 100 polls per second) in order to see if something happened that second that was relevant to them?) Not a problem for MySQL, that's what it does best. The astguiclient application can do 20+ queries per second per client depending on the application. I currently have one company with over 200 client applications(AJAX) sending 3000-4000 queries per second to the MySQL server, and it handles it just fine. On the client side, the load is not very high either, even on a PIII 700MHz machine. Nice. And using a DB to cache events no doubt simplifies the mechanics of the application making it easier to develop and maintain. -- Warm Regards, Lee ___ --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
Re: [asterisk-users] asterisk manager interface stability
Dean Collins wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, 18 May 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk manager interface stability Neat. So the clients use a polling model? Individual clients then query only for events that are interesting? Warm Regards, Lee Yes, the clients only connect to the MySQL database and can query the events as they need to for their display. MATT--- ___ Wouldn't that make it highly inefficient? Is there no two way dialog or am I missing something? Basically if I have 100 end user clients that needed real time interaction with astproxy are you saying that each client would need to poll once per second (eg 100 polls per second) in order to see if something happened that second that was relevant to them?) Although I would lean toward an in-memory cache/handling of events, you could have a another app or thread pool that queries the database on behalf of the clients and notifies clients accordingly, which might negate the need of clients to poll the database and reduce network traffic. -- Warm Regards, Lee ___ --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
