RE: [Asterisk-Users] 407 Proxy Authentication Required
Title: 407 Proxy Authentication Required We also have the same problem over long latency networks ATA also gives Call Rejected: 407. We have tried a lot of different phones and soft phones and the only one working is Xten. In any case this is apparently only problem with newer versions of * - you can use very old version you can avoided the problem. We were not yet able to find final solution for this problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahan Kalutanthri Sent: Monday, June 13, 2005 3:20 AM To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] 407 Proxy Authentication Required I am getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network using sjphone snom. How to overcome this..!! Pls advice..! Shahan This e-mail may contain confidential and/or privileged information. If you are not the intended recipient or have received this e-mail in error, please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-Way Calling in Asterisk
Is it possible to have simple 3-way calling in Asterisk without moving the call to conference room? I was not able to find a way of doing it. Has someone done this? Thanks, Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA; however X-Pro is having no problem. Similarly if the user is not authenticated - the call goes through fine. Is * timing out waiting for some response - if so how can we increase the timeout? We are also thinking of possibility that some port is closed over the long latency network that causing this problem - is that possible -if yes, which port would that be? Or is there another issue that we are not aware of? Thanks, Aram Debug (some parts of it) === Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 360 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp From: 2000sip:[EMAIL PROTECTED];tag=608598751280 CSeq: 1 INVITE Max-Forwards: 70 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f7 9bb0045 User-Agent: SJLabs-SJphone/1.40.258 = 12 headers, 16 lines Ignoring this request Transmitting (no NAT): SIP/2.0 503 Unavailable Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f00244237ee6052da000 00047 From: 2000sip:[EMAIL PROTECTED];tag=608598751280 To: sip:[EMAIL PROTECTED];tag=as0338b9e1 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 = toyyy.yyy.yyy.yyy:5060 In the log Mar 18 07:32:37 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel Mar 18 07:32:38 DEBUG[2325]: Stopping retransmission on '[EMAIL PROTECTED] Mar 18 07:32:38 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 DEBUG[2325]: Ignoring too old packet packet 1 (expecting = 2) Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA; however X-Pro is having no problem. Similarly if the user is not authenticated - the call goes through fine. Is * timing out waiting for some response - if so how can we increase the timeout? We are also thinking of possibility that some port is closed over the long latency network that causing this problem - is that possible -if yes, which port would that be? Or is there another issue that we are not aware of? Thanks, Aram Debug (some parts of it) === Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 360 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp From: 2000sip:[EMAIL PROTECTED];tag=608598751280 CSeq: 1 INVITE Max-Forwards: 70 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f7 9bb0045 User-Agent: SJLabs-SJphone/1.40.258 = 12 headers, 16 lines Ignoring this request Transmitting (no NAT): SIP/2.0 503 Unavailable Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f00244237ee6052da000 00047 From: 2000sip:[EMAIL PROTECTED];tag=608598751280 To: sip:[EMAIL PROTECTED];tag=as0338b9e1 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 = toyyy.yyy.yyy.yyy:5060 In the log Mar 18 07:32:37 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel Mar 18 07:32:38 DEBUG[2325]: Stopping retransmission on '[EMAIL PROTECTED] Mar 18 07:32:38 DEBUG[2325]: Setting NAT on RTP to 4 Mar 18 07:32:38 DEBUG[2325]: Ignoring too old packet packet 1 (expecting = 2) Mar 18 07:32:38 NOTICE[2325]: Unable to create/find channel attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering Machine Detection with app_machinedetect.c
Hello, Is there any documentation available on how to use app_machinedetect.c to detect answering machine? Or is there anyone who can give me some pointers? We have compiled * with app_machinedetect.c, but not able to use it correctly in our configuration. Thanks, Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to monitor Agen Voice channal?
Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in conference room - after the customer hangs up the agent session stays connected and there is no way to disconnect agent session but to restart Asterisk - is this a know problem? Is there a solution for this? But in any case if possible to monitor voice channel of the agent without placing them in conference room we will prefer to use that option. Thank you in advance for help. Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to monitor Agen Voice channal?
So you have to have a Zap device for that. Aram -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Thursday, February 24, 2005 1:59 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to monitor Agen Voice channal? I know you can do this on Zap lines using Zapbarge... Ive tested it and works great...! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aram Ter-Martirosyan Sent: Jueves, 24 de Febrero de 2005 03:50 p.m. To: 'Asterisk Developers Mailing List'; asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to monitor Agen Voice channal? Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in conference room - after the customer hangs up the agent session stays connected and there is no way to disconnect agent session but to restart Asterisk - is this a know problem? Is there a solution for this? But in any case if possible to monitor voice channel of the agent without placing them in conference room we will prefer to use that option. Thank you in advance for help. Aram Ter-Martirosyan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring an active call
Hello, Would it be possible to monitor an extension in asterisk real time (not record and than monitor). To call an extension on asterisk and be able to monitor specific extensions, by punching in that extension number, maybe a password too (for training purpose). The calls are not in conference group, just a regular call from extension to outside number. If yes how can we set it up? Thanks in advance Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring an active call
Hello, Would it be possible to monitor an extension in asterisk real time (not record and than monitor). To call an extension on asterisk and be able to monitor specific extensions, by punching in that extension number, maybe a password too (for training purpose). The calls are not in conference group, just a regular call from extension to outside number. If yes how can we set it up? Thanks in advance Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1800 number with colo
We can give you 800 and local access numbers in US and Canada VoIP (SIP and H.323). All you need is a good internet connection, or you can collocate with us if you like. Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Fraizer Sent: Thursday, July 01, 2004 8:12 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 1800 number with colo Hariharan Gopalan wrote: Hi all Was wondering if anyone is aware of a colo provider who can terminate a 1800 phone line to my box in their colo. I just need one or may be two phone lines with the same 1800 number to go to my asterisk box. Thanks for any help Hariom - Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! We can do this at EnterZone. http://www.enterzone.net/ John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP REGISTER
We have many Asterisk systems connected to us - we provider them with worldwide origination and termination and we have no problems. It could be provider configuration. If you can find out what kind of GW is in use at provider and version I can probably tell you how to configure it properly. Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Micke Andersson Sent: Tuesday, February 17, 2004 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP REGISTER Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -snip SEND provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number sip:[EMAIL PROTECTED] To: pstn-number sip:[EMAIL PROTECTED] Contact: pstn-number sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a2e c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6e9 4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE provider.ip.ip.ip:5060 SIP/2.0 200 OK - end snip - This is Asterisk (does not work) --snip Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: sip:[EMAIL PROTECTED];tag=as017cdd56 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says From: number [EMAIL PROTECTED] asterisk only says From: [EMAIL PROTECTED] How do I tell my Asterisk to send the registration as xlite ? /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat
[Asterisk-Users] RE: IP-PSTN / PSTN-IP Gateway Service Providers
We provide SIP and H323 origination and termination worldwide. Extremely competitive rates. We have may Asterisk clents. Thanks, Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com http://www.hi-teck.com/ 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: Chad Brown [mailto:[EMAIL PROTECTED] Behalf Of Chad Brown Sent: Friday, May 14, 2004 11:27 PM To: [EMAIL PROTECTED] Subject: IP-PSTN / PSTN-IP Gateway Service Providers We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad attachment: winmail.dat
RE: [Asterisk-Users] 1800 Provider
We can provide you 2.2 cents a minute 1800 number through SIP or H.323. We can also provide local access numbers and great worldwide termination rates. Regards, Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jim OnnetSent: Saturday, May 08, 2004 1:11 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] 1800 Provider Hi list, I'm interested inreceiving incoming call to myAsterisk PBXthru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitationsuch asmuch few states that a call can originate? How do the phone cards company with 2cents/minute rate do it by giving out 1800 access number? TIA! Jim
[Asterisk-Users] Need Origination Number form Bahamas
Need SIP or H.323 origination from Bahamas ASAP. Can someone provide an access number or Toll Free number origination from Bahamas? Thanks, Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] California DID Access
We can provide DID in all over southern California. Thanks, Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of SW Sent: Sunday, January 18, 2004 7:44 PM To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] California DID Access Hi, I am looking for a DID access provider in the west cost (not Iconnect, voiceglo, packet8 etc). What I need someone like NuFone, IAX and Toll/Tolfree multiple presentations possible. Any references appreciated. Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Latest version of asterisk
Hello Matt, Is that the Wildcard TE410P you are using. Digium said that it had some problems with Redhat 9.0 is that correct? - Digium quad T1 card - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local) - Redhat 9.0 Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Monday, January 19, 2004 6:21 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Hello, Our max for a single machine is 40 concurrent SIP - Zap conversations for about a 12 hour period and over 5000 total phone calls per day. We didn't see crashes going over that, but we wanted to be safe and now have 2 identical machines handling upto about 30 concurrent SIP - Zap calls(3000 phone calls per day), and a third old machine for office use that never gets over 10 concurrent calls. Here's the specs for these systems: - 120 installed hardphones: - 80 x grandstream 102 hardphones - 20 x Sipura analog adapters(2 phones each) - 2 x Asterisk servers - 2.6 GHz Pentium4 800MHz bus w/ HyperThreading enabled - Asus p4c800 800MHz mobo - 2GB DDR400 RAM (This is actually overkill you need 1GB max if you reboot weekly) - 4 x 36GB SCSI drives in RAID 10 w/megaraid card - 3com 905CX ethernet card - Digium quad T1 card - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local) - Redhat 9.0 - Asterisk with many modules turned off and no MOH With these servers you can see the load average jump from 0.00 to 6.25 in a matter of a minute and then back down again, all while never dropping a call or crashing. We also recently diagnosed our lock-freeze to the touchy manager interface(if you are logged into the manager interface and you loose connection, the manager outgoing buffer seems to overflow and freeze Asterisk). So it doesn't seem to be a problem of hardware. But we still haven't figured out how to fix it. One note as to Ethernet cards, we actually fried a Realtek 8139 Ethernet card that we had put in a server temporarily as we were doing our testing. It started to generate a lot of errors and dropping packets left and right. When we took it out it was VERY hot. We then put in a 3com 905 card and haven't had an issue with it yet. Hope this helps, MATT--- -Original Message- From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 4:49 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Thanks, Matt ! So, am I correct in assuming that there are quite a few (or alot) of us who have had not so good experiences with Asterisk? That Asterisk would crash after it hit a certain number of calls or after a certain period of time with 15-20 calls? I understand that there were others who were able to send a good number of calls through but can anyone tell us if they have had tested and confirmed that Asterisk runs better without or with HT and in terms of number of calls, how many would each one support, in the ballpark? It would also be nice if one could tell us the computer configuration in order to send that many calls without crashing Asterisk. Does it make a difference running the LAN on a ONBOARD LAN card as compared to a PCI Intel or 3COM LAN card, since there is a chance that packets are passing more efficiently on a PCI LAN card? Side question: Is it possible to do passthrough faxing? Like, customers sending me H323 or SIP fax calls and the Asterisk will pass through to another gateway? Anyone successful in doing that? Tommy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Monday, January 19, 2004 8:32 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Hello, I've had Asterisk installed on HT capable machines in both HT mode(with SMP) and non HT mode (with non-SMP) and did not notice any differences functionally between them. The processor load was always less in HT SMP mode than non HT and I have experienced Asterisk deadlocks in both modes so it doesn't really seem to matter if you leave HT on(at least in my experiences). HT basically works by splitting off commands to one of two different virtual processors that both run at about 70% of processor's speed(that's why you may notice compiling to take longer when in HT mode) I have heard of some applications having memory addressing errors with HT but I have not seen any evidence to support that in Asterisk thus far. I'm going to try installing a 4 x T1 card on my Athlon 2xMP server next week and see if Asterisk/Digium performance/compatibility improves over the Intel platform. MATT--- -Original
RE: [Asterisk-Users] ultra-cheap asterisk box
I don't need to use the lowest end server for asterisk, but something reasonable. I need to put one TE410P and 2 TDM400P boards in it. We have about 20 heavy telephone users in the company. Can someone suggest reasonable priced and reliable box? Thanks, Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Thompson Sent: Friday, January 16, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ultra-cheap asterisk box FAQ for Dell 400SC: http://www.aaltonen.us/forums/viewtopic.php?t=8 Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: calvis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 16, 2004 10:33 AM Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I got in on the same Dell deal I think. You must hang out on the bargain boards just like I do? I hang out mainly at fatwallet.com. This is the thread that I got in on the Dell machines that I just recently purchased. http://www.fatwallet.com/forums/messageview.php?start=920catid=24threadid= 264777 I found out by another 400SC user and you can not control assign interrupts on the PCI slots on this machine. Does that point bother you if you are going to run this unit with *? I want to put 3 X100P cards and 1 TDM400P in my up coming 400SC, but not sure if I will have conflict if I use up all the PCI slots in the machine. Charles Alvis Internet Technology Group Redmond, WA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Thursday, January 15, 2004 4:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a 2.4GHz P4 with a 533 MHz front side bus, a 40GB disk, 128MB of memory, sound card, ethernet, and year of on-site next day maintenance. It is $318 delivered after rebates. Yes, $318. This is a real server, by the way, not a desktop machine. It also makes NO noise. I can't hear a thing with my ear right next to it. Why would you even THINK about getting anything else? Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson Sent: Thursday, January 15, 2004 9:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ultra-cheap asterisk box I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means low total cost of ownership To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy
[Asterisk-Users] (no subject)
We are new in Asterisk - I was wondering if someone can recommend a good phone sets to use with Asterisk in office environment. We need about 20 sets. Also - What can we use for the receptionist phone? Thanks, Aram Ter-Martirosyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] We are thinking of Asterisk
Hi all, We are thinking of changing our Nortel Meridian PBX to Asterisk. Before we jump into this we would like to know if we can support some important for us functionalities on Asterisk. We would like to know if we can 1. Have menu based voice mail with Asterisk? (like press 1 for sales, press 2 for tech support ...) 2. If we can login, logout make ourselves unavailable with Asterisk ACD? If Asterisk will route alls FIFO bases to the loggedin and available personal. 3. Can we have regular PBX functionalities Call Forwarding Call Transfer Conference Call Music on Hold 4. Can we point DID directly to the extension? No need to dial a main number and than an extension. 5. Is it possible to monitor a call in Asterisk for training or coaching purposes. If a new person is doing a tech support can we monitor (listen to conversation)? If yes, would it be possible to record conversation - like a voicemail? 6. Can we get our voicemail in our e-mail mailbox? (Can we delete, make new, forward voicemail from e-mail client? 7 Can we send and receive faxes from PC? Thank you in advance for taking time and answering our questions. Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
