[Asterisk-Users] David Choo/eServices/eSpore is overseas
I will be out of the office starting 12/06/2006 and will not return until 17/06/2006. Dear Sir / Mdm, I'm currently travelling. During this period of time, I have minimal access to internet and email. As such, please be aware that I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. For General Technical Queries, please contact Mr Tony Chew @ (65) 6842 2725, Option 2 For VoIP Technical Queries, please contact Mr Randy Khor @ (65) 9800 8468 For Sales Related Queries, please contact our Sales Hotline @ (65) 6842 2725, Option 1 Should you wish to reach me urgently, please contact me @ (65) 6842 2725, Ext - 404 instead. Alternatively, you might wish to drop me a SMS at (65) 90062645 and I will get back to you once I get it. ___ --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] Traffic prioritization and 'class of service' for SIP
Hi Best Regards == David Choo Sales Engineer Citrix Certified Administrator Polycom Qualified Tech Sales Rep Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 DID: 65-9006 2645 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Philip Edelbrock [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 14/02/2006 07:23 AM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To asterisk-users@lists.digium.com cc Subject [Asterisk-Users] Traffic prioritization and 'class of service' forSIP We're got a T1 from Sprint that we use for internet. During VIOP calls, if you download something, the VOIP calls break up. I found some info at Sprint for adding 'class of service', and I also have some information on configuring our Cisco routers. I've read the relevent pages on the wiki, but it seems vauge what's required and what's required by the NSP (Sprint). How have you dealt with this problem? Is this something which requires the NSP to be involved, or can this all be done on the premises side in IOS configuration(s)? Phil ___ --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] David Choo/eServices/eSpore is overseas
I will be out of the office starting 29/10/2005 and will not return until 13/11/2005. Dear Sir / Mdm, I'm currently out of office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please contact the Espore Technical Support Hotline at +65-68422725 and select option 2. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID PHP Script
Hi All, As far as I'm aware, there is this PHP Script that allows us to add / remove callerID from Asterisk's Database? However, as my HDD crashed, I'm unable to search back my old archives. Would anyone be kind enough to point me to the correct URL? Thanks. Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] David Choo/eServices/eSpore is overseas
I will be out of the office starting 10/10/2005 and will not return until 15/10/2005. Dear Sir / Mdm, I'm currently on a Overseas Business Trip. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please contact the Espore Technical Support Hotline at +65-68422725 and select option 2. However, during this period of time, I'm still contacted via my Mobile Phone. Please feel free to contact me should you feel necessary. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] David Choo/eServices/eSpore is overseas
I will be out of the office starting 12/09/2005 and will not return until 16/09/2005. Dear Sir / Mdm, I'm currently on course and are not in office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please contact the Espore Technical Support Hotline at +65-68422725 and select option 2. However, during this period of time, I'm still contacted via my Mobile Phone. Please feel free to contact me should you feel necessary. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly Problem
Hi All, I'm facing a very funny situtation when dealing with Firefly. When the firefly extensions are being dialed, Firefly will hear 1 ring, before hearing the called party's voice, all while the called party is hearing the dialing tones. When Firefly picks up the calls accordingly, the calls will be able to go through like normal, but * don't seem to detect that the called has gone through. After 20 seconds, the calls will be dropped for some reasons. As though its not correct. Do note that it don't seem to be a protocol problem, as IAXComm don't have this issue. Here is the iax debug Start IAX Debug = Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 0ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] FORMAT : 2 asterisk*CLI -- Call accepted by 202.156.XXX.XXX (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00062ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 0ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00066ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: VOICE Subclass: 2 Timestamp: 00080ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 002 Type: VOICE Subclass: 2 Timestamp: 00080ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00080ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 04892ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 04892ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: VOICE Subclass: 2 Timestamp: 04941ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 04941ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: LAGRQ Timestamp: 10032ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10032ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10032ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: PING Timestamp: 20021ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: LAGRQ Timestamp: 20024ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: PONG Timestamp: 20021ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 20021ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 005 Type: IAX Subclass: LAGRP Timestamp: 20024ms SCall: 00042 DCall: 2 [202.156.XXX.XXX:4569] Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass: ACK Timestamp: 20024ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] -- Nobody picked up in 2 ms -- Hungup 'IAX2/892-2' Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 007 Type: IAX Subclass: HANGUP -- Executing Goto(Zap/2-1, s-NOANSWER|1) in new stack Timestamp: 21081ms SCall: 2 DCall: 00042 [202.156.XXX.XXX:4569] -- Goto (macro-stdexten,s-NOANSWER,1) CAUSE CODE : 0 End IAX Debug = Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet
[Asterisk-Users] Answered but No Answer
Dear All, We've successfully implemented * in my office. Everything seems to work fine, but I noticed 1 issues. Calls to certain numbers (800 / 1800 free calls in general) will show that the system detect it as NO ANSWER, even if they are actually picked up. Anyone has any ideas? Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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: **POSSIBLE SPAM** [Asterisk-Users] AreskiCC - DOES IT REALLY WORK??????
I think we should be thankful that the authors are relasing the software, rather then crying out loud when you cannot get it to work. More people will be willing to help you that way. Be ashamed of yourself! Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. [EMAIL PROTECTED] t.com Sent by: To asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject **POSSIBLE SPAM** [Asterisk-Users] 31/05/2005 11:26 AreskiCC - DOES IT REALLY PMWORK?? Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi all, I am quite disappointed at the application AreskiCC. I have installed everything following the instructions but the thing doesnt want to work. First of all, when I start the index.php page, any name/password logs in. After the login it takes me to a page with a single option LOGOUT We are monitoring the database and it seems like the application doesnt connect to it. Does anybody in this have made this work? Can someone help me please?? Thanks, Robson___ 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
RE: [Asterisk-Users] Asterisk on Soekris
Colin is spot on. Couple of years back, we were using Deskpro with 256MB to Run Win2K Server and Lotus Domino with 10 clients. It ran flawlessly until we moved to a proper server. The same Lotus Domino Server, with more RAM, is now my Linux Firewall with over 10 Vlans and around 500 hosts behind it today. Only goes down when me or my colleagues decide to do something funny with it. Hail Compaq... too bad HP didn't do too good a job at integrating... Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Colin Anderson [EMAIL PROTECTED] asterbuilder.com To Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 31/05/2005 10:13 RE: [Asterisk-Users] Asterisk on PMSoekris Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com So, I'm wondering does anyone have real-life comparisons on the failure rate of a PC compared to the failure rate of some of these options?? Obviously, an embedded PC or something that is designed such as a Sokeris is made to last a *long* time, but in my experience, a Tier 1 PC (older Compaq, HP, *not* consumer) PC fares well. I use old Tier 1 PC's for utility jobs like small firewalls or FTP servers or hell even homebrew SAN's and the like, and they just keep chugging. I've never seen a power supply die on a Deskpro, and I've been using them for 10 years. They seem immune to the stupid minor problems that bring clones to a halt, like dust in the fans. I'd never use a clone in an an application where the life expectancy is greater than a year. I sleep well at night knowing that all of those old PC's will be quietly running and doing their jobs just fine the next day. Also, Tier 1 PC's typically are well documented, you can still get drivers for them, and the design is consitient and *made* for business applications. For example, every Deskpro ever made allows you to run it headless, there's an option for it right in the BIOS. ___ 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] Zaptel on pSeries
Dear All, Has anyone got the Digium range of cards working on pSeries? The card seems to be able to be detected by kudzu (I'm running RHEL3.0AS) btw, but the drivers can't be compiled... Asterisk and libpri worked fine though. Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 SIP: [EMAIL PROTECTED] E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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] Connecting 20+ asterisk servers together
Actually, this is whats facing me right now. I think Dundi will resolve the problem, but I've never really placed it to the test. Anyone tested Dundi? Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 SIP: [EMAIL PROTECTED] E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Alex Barnes [EMAIL PROTECTED] software.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 09/05/2005 10:26 RE: [Asterisk-Users] Connecting 20+ PMasterisk servers together Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com -Original Message- From: Vikram Rangnekar [mailto:[EMAIL PROTECTED] Sent: 09 May 2005 11:26 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Connecting 20+ asterisk servers together I have 20+ asterisk servers and need to network them together so a phone on any of the servers can call a phone on any other server without any trouble. I can think of IAX trunks between every server. So every server will have an IAX trunk to every server and then prefix bases routing in the dialplan for each server (I can give a number to each server and use that as a prefix for that server). But I think this is a maintainance nightmare and also a very bad approch does anyone have any better ideas, Also should the phones be able to send rtp between each other or only through the Asterisk server since if its through the asterisk server and say an IAX trunk then the max number of calls can be controlled right. Can dundi or the switch statement help me get out of this mess ? Am I right in saying this is for remote sites rather than 20 servers for load balancing reasons? As I am about to start hooking up a small number ( 5) sites together with Asterisk servers at each site and am not entirely sure of the best approach. I was thinking of having extensions for each site something like: 1xxx = site 1 2xxx = site 2 then for example server1 would have: switch = IAX2/user:[EMAIL PROTECTED]/context Matching on 2xxx. But this doesn't sound particularly elegant specially once you start trying to scale it. If you do get any other ideas I would be interested to know so that I can start this structure out properly. Again if this is remote sites, how are the phones going to talk directly to each other, VPN? Passing the RTP data over VPN direct to the phones will mean you don't get the benefits of the IAX trunking to reduce bandwidth which would be a shame. I would be interested to know how people find VPN's for passing audio, specially if IPSec etc is being used. I would imagine the quality is fairly bad
Re: [Asterisk-Users] Put a wait in a .call file.
John, Since you think its a serious shortcoming, either you fix it or you shut up. To start bitching here and complain that its considered and not implemented is bullshit. * is a great product, but all great product has their flaws. Being OSS, you can always modify the code yourself. Otherwise just ask nicely and someone probably wouldn't mind helping. Best Regards, David Choo John Novack [EMAIL PROTECTED] g-carlson.org To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 04/05/2005 09:48 Re: [Asterisk-Users] Put a wait in PMa .call file. Please respond to [EMAIL PROTECTED] -carlson.org; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Ronan Eckelberry wrote: Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. Thanks, -Ronan This seems to be a serious shortcoming in Asterisk. Can anyone explain why listening for dialtone wasn't an early consideration? With all the toneplans , by country, that are defined, it seems this was considered, but then never made to work John Novack ___ 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] Firefly Qualify Problem
Dear All, I'm using CVS-HEAD 06/04/05 with Realtime, and at present, its working fine generally. However, I'm facing a problem that I find it strange and would like to seek your kind advise. I'm using Firefly 1.9.8 build 3945 and I realise that when I set qualify to yes, then then Asterisk will qualify me as UNREACHEABLE. However, choosing not to qualify will work properly. Is there anyway I can resolve this? For some reason I cannot use IAX, so thats out. Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 SIP: [EMAIL PROTECTED] E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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] Recommended Linux Dist. for Asterisk
We used gentoo internally. I also have * running on CentOS, RHEL. Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Michael George [EMAIL PROTECTED] a.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 21/04/2005 10:31 Re: [Asterisk-Users] Recommended PMLinux Dist. for Asterisk Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Wed, Apr 20, 2005 at 10:26:33PM -0500, Paul Shiflet wrote: I'm trying to find out what flavor of Linux people are choosing for their asterisk boxes. I have been using RH, but i'd like to try some different ones. It seems that RH is the common denominator in this rash of line noise problems. So some suggestions for what dist to use would be great. We use gentoo. Many people would not go that route, but we use that on our servers because when we are ready to update it, we can do so with less pain than with RHL/Fedora and SuSE, etc. The updates of the latter usually go okay, but there comes the time when we need to change major releases and that should be done with a clean reinstall. Now, with * you don't really need to do any changing as it will just sit there and work for the most part. However, since we have gentoo in many of our systems, we just stick with that. The ports in gentoo stay pretty current and it's worked fine for us. YMMV, and as I said above, gentoo is probably not the route for many who have little linux experience. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Wait in Dial String
Dear All,
My boss has placed a requirement for me to forward all our IDD calls
through a partner's IDD service, which requires us to call a 8 digit
number, wait for 1 sec, before we send in the foreign number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm only removing the
1st 3 digits (900) and sending in an extra 1 to simulate the wait. It
works, but not all the time. Is there anyway that I can place a wait
command here? I'm tried placing w / p but both don't works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
Privileged/Confidential information may be contained in this message. If
you are not the intended recipient, you must not copy it or use it for any
purpose, nor deliver this message to anyone. Instead, please delete this
message and destroy any other record of it immediately and kindly notify
the sender by return email. Thank you for your co-operation.
Internet communications cannot be guaranteed to be secure or error-free as
information could be intercepted, corrupted, lost, arrive late, or contain
viruses. The sender therefore does not accept liability for any errors or
omissions in the context of this message nor can the sender guarantee that
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Re: [Asterisk-Users] Wait in Dial String
Guys,
Thanks a mil. I'll try it out and see how!
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
Privileged/Confidential information may be contained in this message. If
you are not the intended recipient, you must not copy it or use it for any
purpose, nor deliver this message to anyone. Instead, please delete this
message and destroy any other record of it immediately and kindly notify
the sender by return email. Thank you for your co-operation.
Internet communications cannot be guaranteed to be secure or error-free as
information could be intercepted, corrupted, lost, arrive late, or contain
viruses. The sender therefore does not accept liability for any errors or
omissions in the context of this message nor can the sender guarantee that
this message is virus free.
Robert Webb
[EMAIL PROTECTED]
u.com To
Sent by: asterisk-users@lists.digium.com,
asterisk-users-bo cc
[EMAIL PROTECTED]
m.com Subject
Re: [Asterisk-Users] Wait in Dial
String
20/04/2005 11:54
PM
Please respond to
Asterisk Users
Mailing List -
Non-Commercial
Discussion
[EMAIL PROTECTED]
ists.digium.com
On Wed, 20 Apr 2005 10:24:37 -0500
Josiah Bryan [EMAIL PROTECTED] wrote:
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all
our IDD calls
through a partner's IDD service, which requires us to
call a 8 digit
number, wait for 1 sec, before we send in the foreign
number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm
only removing the
1st 3 digits (900) and sending in an extra 1 to simulate
the wait. It
works, but not all the time. Is there anyway that I can
place a wait
command here? I'm tried placing w / p but both don't
works. Would like to
seek your kind assistance!
exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
exten = _9001.,n,Hangup()
Try 'w',
E.g. for my old bridge to BizFon, I had to dial 9, wait,
then the number:
exten = _NX,1,Dial(Zap/g1/9w${EXTEN})
Just put the 'w' between the numbers that you want it to
'wait' at.
-josiah
And as an added tidbit... If I remeber correctly, each w
is about a 1/2 second. So to get a second pause you would
need ww in the string.
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[Asterisk-Users] Polycom V500 With Asterisk Setup
Dear All, We've got 1 set of Polycom V500 Video Conferencing Kit in my Office. I'm trying to link it with Asterisk and is facing some issues. Would like to seek your kind advise. The Polycom V500 is unable to make the outgoing calls, and will always report the ENTER ERROR HERE. sip show peers does not shows that the Polycom V500 being able to register. The account is working alright as I've used the account on Eyebeam and its working fine. Here are the debug logs for the System -- SIP read from 192.168.100.146:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];epid=82042503E72EB0;tag=df8c4526 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Polycom V500 Release 7.5 - 15Dec2004 10:12 Contact: sip:192.168.100.146 Content-Type: application/sdp Content-Length: 899 v=0 o=Vigor11 1627471320 0 IN IP4 192.168.100.146 s=- c=IN IP4 192.168.100.146 b=AS:384 t=0 0 m=audio 49178 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:98 SIREN14/16000 a=fmtp:98 bitrate=32000 a=rtpmap:97 SIREN14/16000 a=fmtp:97 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 m=video 49180 RTP/AVP 109 34 96 31 b=TIAS:384000 a=rtpmap:109 H264/9 a=fmtp:109 profile-level-id=42800c max-mbps=1 a=rtpmap:34 H263/9 a=rtpmap:96 H263-1998/9 a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 F J T a=rtpmap:31 H261/9 a=fmtp:31 CIF=1 QCIF=1 m=data 49182 RTP/AVP 100 a=rtpmap:100 H224 --- (11 headers 35 lines)--- Using latest request as basis request Sending to 192.168.100.146 : 5060 (non-NAT) Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5947 check_user_full: Setting NAT on RTP to 524288 Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5951 check_user_full: Setting NAT on VRTP to 524288 Reliably Transmitting (NAT) to 192.168.100.146:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655;received=192.168.100.146;rport=5060 From: sip:[EMAIL PROTECTED];epid=82042503E72EB0;tag=df8c4526 To: sip:[EMAIL PROTECTED];tag=as36644353 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: nVoice PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=nvoice, nonce=60b31ab3 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '898' tannery*CLI -- SIP read from 192.168.100.146:5060: ACK sip:192.168.100.146 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];epid=82042503E72EB0;tag=df8c4526 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Contact: sip:192.168.100.146 Content-Length: 0 Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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] Direct Channel Answering
Dear Cameron,
Thanks! That really kicked me off into achieving what i desired.
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
Privileged/Confidential information may be contained in this message. If
you are not the intended recipient, you must not copy it or use it for any
purpose, nor deliver this message to anyone. Instead, please delete this
message and destroy any other record of it immediately and kindly notify
the sender by return email. Thank you for your co-operation.
Internet communications cannot be guaranteed to be secure or error-free as
information could be intercepted, corrupted, lost, arrive late, or contain
viruses. The sender therefore does not accept liability for any errors or
omissions in the context of this message nor can the sender guarantee that
this message is virus free.
Cameron Beattie
[EMAIL PROTECTED]
nzTo
Sent by: Asterisk Users Mailing List -
asterisk-users-bo Non-Commercial Discussion
[EMAIL PROTECTED] asterisk-users@lists.digium.com
m.com cc
Subject
07/04/2005 01:52 Re: [Asterisk-Users] Direct Channel
PMAnswering
Please respond to
Cameron Beattie
[EMAIL PROTECTED]
m.com; Please
respond to
Asterisk Users
Mailing List -
Non-Commercial
Discussion
[EMAIL PROTECTED]
ists.digium.com
In zapata.conf you can associate the channel with a context. So create
different contexts for each channel. I s'pose you can do that for PRI or
whatever.
Regards
Cameron
- Original Message -
From: David Choo [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 2:42 PM
Subject: [Asterisk-Users] Direct Channel Answering
Dear All,
I'm trying to achieve the following with asterisk, would like to seek
your
kind advise.
1: All calls to Channel 8 will go to extension 112
2: All calls to Channel 12 will go to extension 113
I've tried to read README.varibles and voip-info. It don't seem to help
much, would just like some kind soul to help me out on this. Thanks!
Here are my extension.conf entries.
[incoming]
exten = s,1,GotoIf($[${CHANNEL:8:1} = 8]?local-extensions,112,1:)
exten = s,n,GotoIf($[${CHANNEL:12:1} = 8]?local-extensions,113,1:)
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
Privileged/Confidential information may be contained in this message. If
you are not the intended recipient, you must not copy it or use it for
any
purpose, nor deliver this message to anyone. Instead, please delete this
message and destroy any other record of it immediately and kindly notify
the sender by return email. Thank you for your co-operation.
Internet communications cannot be guaranteed to be secure or error-free
as
information could be intercepted, corrupted, lost, arrive late, or
contain
viruses. The sender therefore does not accept liability
Re: [Asterisk-Users] Access Voicemail From Outside
Hi, Try this. What it does is to create a extention 250 which you can call from outside. Thats what I do in my office. Replace 250 with your preferred number. exten = 250,1,Answer() exten = 250,n,VoicemailMain() exten = 250,n,Hangup() Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Bill Ford [EMAIL PROTECTED] .com To Sent by: [EMAIL PROTECTED] [EMAIL PROTECTED] asterisk-users-bo cc [EMAIL PROTECTED] Asterisk Users Mailing List - m.com Non-Commercial Discussion asterisk-users@lists.digium.com Subject 07/04/2005 04:09 Re: [Asterisk-Users] Access PMVoicemail From Outside Please respond to Bill Ford [EMAIL PROTECTED] .com; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I've tried disa, but haven't been very successful. Perhaps I'm missing something in the configuration. But what I'd really prefer is limiting the acccess to just voicemail. On Apr 7, 2005 3:05 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You can use a password protected DISA functionality. Selon Bill Ford [EMAIL PROTECTED]: I'd like to see what some of you are doing to reliably aess voicemail from an outside line. Thanks Bill ___ 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
Re: [Asterisk-Users] Automatic Start
Hi, If its started from an automated script, it should be in /etc/rc.d/init.d/asterisk If you want a quick and easy fix to start asterisk after zaptel do this (Assuming that Zaptel is installed properly) Modify /etc/rc.local Add the following lines. service asterisk stop service zaptel restart service asterisk start It will first start asterisk, then restart zaptel (to ensure that even if its started, it will work) then start asterisk Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. J Hobbs [EMAIL PROTECTED] il.comTo Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Automatic Start 06/04/2005 08:17 AM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi All, I have just installed Asterisk using Signate install disks which installed CentOS (Redhat) and then loaded Asterisk. I am completely new to Linux so this is probably a dumb question, but I can't figure it out or find any documents that refer to it. So here goes. When I start CentOS it automatically loads Asterisk and Gnome, but where is the startup configuration file for Asterisk. I need to edit this file so that the Wildcard is also loaded. Put me out of my misery please. _ Powerful Parental Controls Let your child discover the best the Internet has to offer. http://join.msn.com/?pgmarket=en-capage=byoa/premxAPID=1994DI=1034SU=http://hotmail.com/encaHL=Market_MSNIS_Taglines Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*. ___ 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
Re: [Asterisk-Users] Queues
Hi Henry, Thanks for the advise. I'll check that out. Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. Henry Devito [EMAIL PROTECTED] m To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 02/04/2005 01:57 Re: [Asterisk-Users] Queues PM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com After calls come in, it works fine, however, I notice that even when SIP/602 is on the phone, Asterisk will still ring her. I believe its due to the fact that the phone support call-waiting. Is there anyway that I can disable this support only on queues and ring the next extension in this case, which is SIP/603? You can use setgroup/checkgroup combo I think. What kind of phone is SIP/602. Most phone have a config to disable call waiting. ___ 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] Queues
Dear All, I've got a working asterisk installation which I need minor help from. Currently, I'm running a Sales Queue, which is answered by a selected group of people. Here are my queues.conf [sales-hotline] strategy = roundrobin timeout = 10 member = SIP/602 member = SIP/603 member = SIP/701 member = SIP/604 After calls come in, it works fine, however, I notice that even when SIP/602 is on the phone, Asterisk will still ring her. I believe its due to the fact that the phone support call-waiting. Is there anyway that I can disable this support only on queues and ring the next extension in this case, which is SIP/603? ringall might be a good workaround to resolve this problem, but i would like to avoid this as it will result in all phones leaving missed calls. Would appreciate any form of advise. Thanks! Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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] Realtime
Dear All, I'm currently running Asterisk Relatime with pgSQL and unixODBC. However, I'm facing a little problem here. Whenever I reloads asterisk, current SIP registration will be lost. Is there anyway that I can retain this SIP registration after reloads, or at least set a timeout for reloads? Forwarded are an example of what I mean. Would appreciate any kind advise. === Pre Reload === asterisk*CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status 501/501192.168.100.153 D N 255.255.255.255 5060 OK (84 ms) 708/708192.168.100.212 D N 255.255.255.255 5060 OK (83 ms) 502/502192.168.100.196 D N 255.255.255.255 5060 OK (79 ms) 103/103192.168.100.192 D N 255.255.255.255 5060 OK (81 ms) 604/604192.168.100.198 D N 255.255.255.255 5060 OK (126 ms) 701/701(Unspecified)D N 255.255.255.255 0 UNKNOWN 603/603192.168.100.203 D N 255.255.255.255 5060 OK (80 ms) 602/602192.168.100.166 D N 255.255.255.255 5060 OK (79 ms) === Pre Reload === === Post Reload === asterisk*CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status 0 sip peers [0 online , 0 offline] asterisk*CLI === Post Reload === Best Regards, == David Choo Systems Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-68487806 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ 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] Voicemail Problems
Dear All,
I've setup got a Asterisk and pgSQL combi that works fine. I'm about to
perform the migration deployment when I noticed a issue which I need some
expert advise here.
When user connect to Voicemail, the CPU Load of the machine will shoot up
to around 50 - 60%, and its causing sound distortion, and not to mention
serious discomfort during my demo. Call to unavailable users will yield the
same result, calls to busy users will yield the same result too.
However, PSTN / IP calls all work smoothly. Similary, my IVR works
perfectly.
I've tried adding, and subsequently removing the following sample lines to
no effect.
exten = s,1,answer()
I might have missed something out, and I don't have much time left. Would
appreciate any help. I'm forwarding only part of the extensions.conf here
as I don't want to jam up the mail, but if anyone requires, please buzz me
and I'll forward you the entire file! Cheers!
= Start
===
exten = a,1,VoicemailMain(${MACRO_EXTEN})
exten = a,n,Hangup()
exten = s,1,NoOp(${ARG1})
exten = s,n,NoOp(${ARG2})
exten = s,n,NoOp(${ARG3})
exten = s,n,NoOp(${ARG4})
exten = s,n,NoOp(${ARG5})
exten = s,n,NoOp(${ARG6})
;exten = s,n,GotoIf($[${CALLERIDNAME} = ]?setName:skipSetName)
;exten = s,n(setName),SetCIDName(${CALLERIDNUM})
exten = s,n,SetCIDName(${CALLERIDNAME})
exten = s,n,SetCIDNum(${CALLERIDNUM})
exten = s,n,GotoIf($[${ARG4} != 0]?${ARG2},${ARG4},1:)
exten = s,n,Dial(SIP/${ARG1}IAX2/${ARG1},${ARG3},,TtWw)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,GotoIf($[${ARG5} != 0]?${ARG2},${ARG5},1:)
exten = s-BUSY,n,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,n,Hangup()
exten = s-NOANSWER,1,Answer()
exten = s-NOANSWER,n,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
=== End
==
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :[EMAIL PROTECTED]
=
Privileged/Confidential information may be contained in this message. If
you are not the intended recipient, you must not copy it or use it for any
purpose, nor deliver this message to anyone. Instead, please delete this
message and destroy any other record of it immediately and kindly notify
the sender by return email. Thank you for your co-operation.
Internet communications cannot be guaranteed to be secure or error-free as
information could be intercepted, corrupted, lost, arrive late, or contain
viruses. The sender therefore does not accept liability for any errors or
omissions in the context of this message nor can the sender guarantee that
this message is virus free.
___
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