Re: [asterisk-users] Snom 320 echo
Hi Mike, since you are using snom phones I have 2 question for you maybe you can help me (no one else could till now): 1 - sometimes the message bad gateway appears on my phone display. 2 - after a call the phone led do not turn off so anyone else does not know if I'm busy or not. Have you ever had problems like these? TIA Giorgio Mike Hammett wrote: Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.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 ___ --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] ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro.
The logic is simple; try Zap/g1 (a group of two E1s), and if that
fails, try locating a channel via DUNDi. Here's a massively cut down version
to illustrate the problem I'm having.
macro dialout ( dest ) {
ChanIsAvail(Zap/g1);
noop(Value of AVAILCHAN is ${AVAILCHAN});
Dial(Zap/G1/${dest}||H);
};
Here's what happens when only the second bearer is connected:
-- Executing Macro(SIP/1210-082a9768, dialout|0800789456) in new stack
-- Executing Set(SIP/1210-082a9768, dest=0800789456) in new stack
-- Executing ChanIsAvail(SIP/1210-082a9768, Zap/g1) in new stack
-- Hungup 'Zap/32-1'
-- Executing NoOp(SIP/1210-082a9768, Value of AVAILCHAN is Zap/32-1) in
new stack
-- Executing Dial(SIP/1210-082a9768, Zap/G1/0800789456||H) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/0800789456
-- Zap/62-1 is making progress passing it to SIP/1210-082a9768
-- Zap/62-1 answered SIP/1210-082a9768
-- Hungup 'Zap/62-1'
== Spawn extension (macro-dialout, s, 4) exited non-zero on
'SIP/1210-082a9768' in macro 'dialout'
== Spawn extension (macro-dialout, s, 4) exited non-zero on
'SIP/1210-082a9768'
i.e. perfect - it picks the first available channel on the second
bearer - Zap/32. If only the first bearer is connected, it picks Zap/1
as I'd expect.
The killer is if /neither/ bearer is connected, I get this:
-- Executing Macro(SIP/1210-08299328, dialout|0800789456) in new stack
-- Executing Set(SIP/1210-08299328, dest=0800789456) in new stack
-- Executing ChanIsAvail(SIP/1210-08299328, Zap/g1) in new stack
== Spawn extension (macro-dialout, s, 3) exited non-zero on
'SIP/1210-08299328' in macro 'dialout'
== Spawn extension (macro-dialout, s, 3) exited non-zero on
'SIP/1210-08299328'
Processing does not continue to the NoOp or Dial - what am I doing
wrong? I've also tried with the 'j' option to 'jump to priority n+101'
even though I'm using AEL, but it's made no difference.
Cheers,
Gavin.
___
--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] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ?
Aren't Aastra due to release new phones and some form of operator/reception addon? The Aastra user/admin guides are a lot more up2date that they used to be. I'd like Aastra to add a GSM codec to their phone and have a more regular firmware release schedule. I agree with the list below though that Polycom does have a better line up currently, and especially point 7 - when rebooting the phone please don't drop the network ports. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: 23 January 2007 02:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ? Here are another $0.02 We too have put in a lot of polycoms and aastras. I agree with a lot of what you noted below...but there are two big strikes against aastra: 1. Firmware bugs. Even some basic functions of the 480i are unusable/unstable due to firmware bugs. The word from support is always wait for the next firmware 2. Poor documentation. Their documentation is out of date and lacking a LOT of critical functions. (eg: Try to setup a hold button on the wireless handset using a config file) We're steering more customers towards polycom now. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, January 22, 2007 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra 480i. Which one to choose ? With over 300 Polycoms, and around 80 Aastra 480i under my belt here is my $0.02. 1. Sound quality, Polycom wins but the Aastra has excellent sound quality as well. 2. Complete product line, Polycom wins. 3. Cordless Aastra, although it's not the best cordless. 4. Backlit, Aastra 5. PoE, Aastra 6. Speakerphone, they both have good speaker phones. Although in general the answer to 1 goes here as well. 7. They both have 2 network ports, but I havnt' done any tests on the speed, I did however notice that when restarting the phone, the Polycom will not shut the network ports down, while the Aastra will. On another note, in general the Polycoms give me less problems. The Aastras are not yet that stable. See my next post to the list. On 1/22/07, Bruce Reeves [EMAIL PROTECTED] wrote: Your list seems to lean heavily to the Aastra, while I choose the Polycom 501/601 over the Aastra, I did like the unit I tested and the cordless. In the end the fact that most of the people using the phones would use the speaker phone, Polycom and their history of conference phones made the choice. We rolled 75 phones at one site and another 30 now at remote locations. As far a a receptionist phone, we choose to use a software operator panel instead of a phone that took up most of the desk, there were initial concerns but the results have been excellent. If you have not already done so grab a few people from different parts of the office and have them give their 2 cents, it will help to have their perspectives on the quality and feel of the phones. On 1/22/07, Vikas [EMAIL PROTECTED] wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --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 -- Bruce Nortex Networks ___ --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 ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/
RE: [asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0
Have you tried the #freepbx IRC channel or the freepbx mailing list? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Arnilo S. Baluyos (Mailing Lists) Sent: 23 January 2007 01:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Music on Hold on IP Phones with FreePBX 2.2.0 Hello everyone, We just installed a new Trixbox 2.0 server and updated FreePBX to 2.2.0 from 2.2.0rc3. We are having some problems with regards to Music on Hold on IP phones. When we press the Hold button, the caller doesn't hear the MOH sound. This functionality used to work with the older [EMAIL PROTECTED] installation on the same hardware and configuration. However, we don't have any problems with softphones only on IP phones. Is there anyone also having the same problem? Best regards, Matt -- Stand before it and there is no beginning. Follow it and there is no end. Stay with the ancient Tao, Move with the present. ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 Failed because: Incorrect information in file: './asterisk/sip.frm'
Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now ! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = 'h' AND context = ' interne' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten = 'h' AND context = 'interne ' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND conte xt = 'interne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'i nterne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/sip.frm 'ast ___ --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 IAX and Shorewall QoS ?
Hi anyone have a sample of shorewall configuration for add a TC/QoS on IAX2 traffic ? Thanks for your help ___ --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] Digium Forums
Hi List, Does anyone know where I can get support for the digium forums ? my user ID and pass just stoped working as of yesterday. The forums say to go to asterisk.org for any password issues. I am able to log in there with out any issues. For some reason when I try to log in to the forums it wont accept it. Anyone have an ideas ? Thanks. Dovid___ --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] beronet DTMF detection problem with some Telecom Italy lines
Ciao, I have an Asterisk 1.2.9.1 box with a beronet dualBRI (install-misdn-queue) on a Debian distro. I'm experiencing problem with some Telecom Italia lines: some people cannot choose menu selections or extensions after hearing intro message. Is there anybody who knows if there is a particular parameter to set inside misdn.conf (or maybe some other configuration files) ? TIA Giorgio Incantalupo ___ --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] NAT
I'm running Asterisk SVN-trunk-r51353... for some reason even if i set nat=yes in the sip.conf for a device when i do a show sip peers it shows N for nat. Is this a bug or am i doing somthing wrong here. I'm basically having a problem right now where i can call in/out of asterisk and talk fine using the phone but if i call another SIP phone registered to the same asterisk server it rings but when i pickup both ends i cant hear or say anything through them. Not sure if this is related to the NAT issue... i think it may be? my setup is... Asterisk is on a public ip 2 polycom601 phones on a private network here's a sip debug... http://channels.debian.net/paste/5164 ~Shaun ___ --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: Can't find asterisk.ctl under CentOS installation
Thanks AT. On 1/24/07, Axel Thimm [EMAIL PROTECTED] wrote: On Tue, Jan 23, 2007 at 06:20:08PM +0100, Axel Thimm wrote: On Tue, Jan 23, 2007 at 03:54:49PM +0200, Tzafrir Cohen wrote: On Tue, Jan 23, 2007 at 02:48:07PM +0100, Axel Thimm wrote: If I call asterisk -r as root it succeeds, if as another user it will give Devraj's error message. That's probably how it is supposed to work, or not? Just a thought: shouldn't the asterisk user be allowed write access to that control socket? Or maybe the asterisk group? The asterisk user is allowed, too, of course, the group not (yet). (for quickdirty shell scripts) I think that makes very much sense. The socket is created by asterisk, is there a parameter to specify permissions/umask of that socket? Looks like all there is needed is to uncomment the following line in the default config file: [files] astctlpermissions = 0660 But since upstream defaults to not do so and only have this done by the user, I wouldn't like to change this policy on the package level. -- Axel.Thimm at ATrpms.net ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] AOC on misdn?
Hi, i can see AOC messages on the asterisk console. Can i sendtext() them to the caller or put them in cdr? Regards, Andreas. _ Need a new job? Check out XtraMSN Careers http://xtramsn.co.nz/careers ___ --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 1.4 Polycom buddy status
I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 23, 2007, at 10:09 AM, Chris Bullock wrote: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --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] vxml support
Can Asterisk support vxml? Can i work with Asterisk and vxml? Is there any AGI framework that can use vxml? 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
[asterisk-users] Re: agi script as member in queue
On 1/22/07, nik600 [EMAIL PROTECTED] wrote: Hi i want to put an AGI script in a queue, to serve once at time the callers. Example: Queue (8 callers waiting) Agi script / IVR (serving the caller) can i do that? Thanks any idea? ___ --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] Grandstream GXP2000 and Interception of call ?
Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone Thanks bye ___ --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] Digium Forums
Dovid B wrote: Hi List, Does anyone know where I can get support for the digium forums ? my user ID and pass just stoped working as of yesterday. The forums say to go to asterisk.org for any password issues. I am able to log in there with out any issues. For some reason when I try to log in to the forums it wont accept it. Anyone have an ideas ? A forum user reported that his user ID got changed. It looks like maybe they merged in the AsteriskNow user/passwords. I know that when I login with the user/password I created to download AsteriskNow, it works for forum access. ___ --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] Panasonic Hybrid Integration Advice Needed
I have a client who has a Panasonic Hybrid system. They are taking in another company as a building tenant and the tenant will be on a new 12 station Asterisk system. This new asterisk system will have 4 FXO ports plus ITSP. The two systems will be separate except that they should tie together for the purposes of dialing extensions directly on the opposite phone system and for transferring calls. I'm looking for advice on how best to accomodate this. Is it possible to do this via the Panasonic's IP interface or will I need to cross connect them via T1 cards? This is my first integration as you can probably surmise. Thanks in advance. -- 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
Re: [asterisk-users] Query Failed because: Incorrect information in file: './asterisk/sip.frm'
Its a problem in your database. something might have corrupted...be prepared to load a backup Gregory Duchatelet wrote: Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now ! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = 'h' AND context = ' interne' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten = 'h' AND context = 'interne ' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND conte xt = 'interne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'i nterne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/sip.frm 'ast ___ --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] RE: Snom 320 echo
Where do I find more out in regards to the echo-cancelling component you mentioned? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 8:08 PM Subject: asterisk-users Digest, Vol 30, Issue 95 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. [OT] Mark Spencer Presents AsteriskNOW on Youtube (Damian Fossi) 2. RE: TDM2400 Hardware Echo Cancel (Webster, Andrew) 3. Re: How to exit from console? (Tzafrir Cohen) 4. OT: High Quality Wireless Headset for Cisco IP Phones and * (Tom) 5. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Cory Andrews) 6. Re: Re: Dial plan constructions suggestions? (Lacy Moore - Aspendora) 7. Re: weird undocumented extensions such as s-BUSY (Barzilai Spinak) 8. Re: TDM2400 Hardware Echo Cancel (Mailing List) 9. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Colin Anderson) 10. Re: Echo... (Matthew Fredrickson) 11. Snom 320 echo (Mike Hammett) 12. RE: Snom 320 echo (Colin Anderson) 13. AW: [asterisk-users] Snom 320 echo (Christian Stredicke) 14. automon and MONITOR_EXEC (John Williams) 15. DB_DELETE Function in 1.4 (Jeremiah Millay) 16. RE: * 1.0.9 Voicemail record name does not playb ack in Directory() --solved (Colin Anderson) 17. Re: How to exit from console? (Paul Hales) 18. Problem connecting PAP2 over wifi bridge (Alfredo Manrique) 19. DeStar 0.2.2 released! (Santiago Jos? Ruano Rinc?n) 20. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Dovid B) 21. Echo on IP phones... (Carlos Chavez) 22. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Kristian Kielhofner) 23. Re: Re: [asterisk-users] How to exit from console? ( ?? ) 24. Re: DB_DELETE Function in 1.4 (Alvin Austin) 25. Re: How to exit from console? (John Novack) 26. Re: realtime sipusers and rtcachefriends... bigheadache!! (kjcsb) 27. cmd Backgound problem with option m (Franz Wu) -- Message: 11 Date: Tue, 23 Jan 2007 15:10:28 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [asterisk-users] Snom 320 echo To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/ef7ed599/attachment-0001.htm -- Message: 12 Date: Tue, 23 Jan 2007 14:21:52 -0700 From: Colin Anderson [EMAIL PROTECTED] Subject: RE: [asterisk-users] Snom 320 echo To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Later firmware versions have an echo-cancelling component in it, upgrade to latest version and also turn down the gains on the mic, the default setting is way too high. A setting of 3 or 4 max is all that is nessisary. hth -Original Message- From: Mike Hammett [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 23, 2007 2:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Snom 320 echo Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/22d41817/attachment-0001.htm -- Message: 13 Date: Tue, 23 Jan 2007 22:30:44 +0100 (MET) From: Christian Stredicke [EMAIL PROTECTED] Subject: AW: [asterisk-users] Snom 320 echo To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
[asterisk-users] RE: Snom 320 echo
Where do I find more out in regards to the echo-cancelling component you mentioned? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 8:08 PM Subject: asterisk-users Digest, Vol 30, Issue 95 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. [OT] Mark Spencer Presents AsteriskNOW on Youtube (Damian Fossi) 2. RE: TDM2400 Hardware Echo Cancel (Webster, Andrew) 3. Re: How to exit from console? (Tzafrir Cohen) 4. OT: High Quality Wireless Headset for Cisco IP Phones and * (Tom) 5. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Cory Andrews) 6. Re: Re: Dial plan constructions suggestions? (Lacy Moore - Aspendora) 7. Re: weird undocumented extensions such as s-BUSY (Barzilai Spinak) 8. Re: TDM2400 Hardware Echo Cancel (Mailing List) 9. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Colin Anderson) 10. Re: Echo... (Matthew Fredrickson) 11. Snom 320 echo (Mike Hammett) 12. RE: Snom 320 echo (Colin Anderson) 13. AW: [asterisk-users] Snom 320 echo (Christian Stredicke) 14. automon and MONITOR_EXEC (John Williams) 15. DB_DELETE Function in 1.4 (Jeremiah Millay) 16. RE: * 1.0.9 Voicemail record name does not playb ack in Directory() --solved (Colin Anderson) 17. Re: How to exit from console? (Paul Hales) 18. Problem connecting PAP2 over wifi bridge (Alfredo Manrique) 19. DeStar 0.2.2 released! (Santiago Jos? Ruano Rinc?n) 20. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Dovid B) 21. Echo on IP phones... (Carlos Chavez) 22. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Kristian Kielhofner) 23. Re: Re: [asterisk-users] How to exit from console? ( ?? ) 24. Re: DB_DELETE Function in 1.4 (Alvin Austin) 25. Re: How to exit from console? (John Novack) 26. Re: realtime sipusers and rtcachefriends... bigheadache!! (kjcsb) 27. cmd Backgound problem with option m (Franz Wu) -- Message: 11 Date: Tue, 23 Jan 2007 15:10:28 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [asterisk-users] Snom 320 echo To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/ef7ed599/attachment-0001.htm -- Message: 12 Date: Tue, 23 Jan 2007 14:21:52 -0700 From: Colin Anderson [EMAIL PROTECTED] Subject: RE: [asterisk-users] Snom 320 echo To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Later firmware versions have an echo-cancelling component in it, upgrade to latest version and also turn down the gains on the mic, the default setting is way too high. A setting of 3 or 4 max is all that is nessisary. hth -Original Message- From: Mike Hammett [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 23, 2007 2:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Snom 320 echo Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/22d41817/attachment-0001.htm -- Message: 13 Date: Tue, 23 Jan 2007 22:30:44 +0100 (MET) From: Christian Stredicke [EMAIL PROTECTED] Subject: AW: [asterisk-users] Snom 320 echo To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
RE: [asterisk-users] RE: Snom 320 echo
Sorry my mistake it is in Snom 360 firmware 3.60b and higher, not 320. -Original Message- From: Mike Hammett [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 24, 2007 7:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE: Snom 320 echo Where do I find more out in regards to the echo-cancelling component you mentioned? ___ --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: Disconnect Supervision UK / BT solution?
Yes, IAX -- IAX works fine it's when it's Zap(in) -Asterisk - Zap(out) Every call that I pass through to that outgoing Zap channel (a dial out to a mobile phone) fails to hang up. CallerID is working -- with this sangoma card, which seems to need... cidsignalling=v23 cidstart=polarity ...to work in the UK If removing helps the disconnect detection, I don't mind losing my callerID support. -- Chris Ed W [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Chris Earle (CBL) wrote: Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS - Sangoma|Asterisk - POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. Just to clarifydoes it all work ok if you are using SIP or IAX for the forwarded channels? Eg local SIP phones? I only have incoming zap lines in my config and with the exception of hangup on ringing I have found hangup detection to work fine. I have a fax machine forwarding in my config as well and again no problems yet with hangup on that. Does it fail to work *every* time, or just intermittently? Does CallerId work ok in your setup? (can be a clue to help diagnose your setup) Ed W ___ --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] Panasonic Hybrid Integration Advice Needed
Which panasonic system? I'm assuming you are talking about the TDA line. If so get a IP Gateway card on the TDA system, that card uses h323, then configure it with asterisk as h323, or my favorite, get a PRI card on the TDA sysem (unless it's a TDA50 then the option is not available), and a T1 card on asterisk, and create a dialplan on the Panasonic that goes out over the PRI card. On 1/24/07, John French [EMAIL PROTECTED] wrote: I have a client who has a Panasonic Hybrid system. They are taking in another company as a building tenant and the tenant will be on a new 12 station Asterisk system. This new asterisk system will have 4 FXO ports plus ITSP. The two systems will be separate except that they should tie together for the purposes of dialing extensions directly on the opposite phone system and for transferring calls. I'm looking for advice on how best to accomodate this. Is it possible to do this via the Panasonic's IP interface or will I need to cross connect them via T1 cards? This is my first integration as you can probably surmise. Thanks in advance. -- 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 ___ --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] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ?
From what I heard: Aastra is coming out with a few new phones 53i, 55i, and 57i and sidecar On 1/24/07, Lee Archer [EMAIL PROTECTED] wrote: Aren't Aastra due to release new phones and some form of operator/reception addon? The Aastra user/admin guides are a lot more up2date that they used to be. I'd like Aastra to add a GSM codec to their phone and have a more regular firmware release schedule. I agree with the list below though that Polycom does have a better line up currently, and especially point 7 - when rebooting the phone please don't drop the network ports. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: 23 January 2007 02:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra480i. Which one to choose ? Here are another $0.02 We too have put in a lot of polycoms and aastras. I agree with a lot of what you noted below...but there are two big strikes against aastra: 1. Firmware bugs. Even some basic functions of the 480i are unusable/unstable due to firmware bugs. The word from support is always wait for the next firmware 2. Poor documentation. Their documentation is out of date and lacking a LOT of critical functions. (eg: Try to setup a hold button on the wireless handset using a config file) We're steering more customers towards polycom now. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, January 22, 2007 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 7 points of comparison Polycom 430/501 andAastra 480i. Which one to choose ? With over 300 Polycoms, and around 80 Aastra 480i under my belt here is my $0.02. 1. Sound quality, Polycom wins but the Aastra has excellent sound quality as well. 2. Complete product line, Polycom wins. 3. Cordless Aastra, although it's not the best cordless. 4. Backlit, Aastra 5. PoE, Aastra 6. Speakerphone, they both have good speaker phones. Although in general the answer to 1 goes here as well. 7. They both have 2 network ports, but I havnt' done any tests on the speed, I did however notice that when restarting the phone, the Polycom will not shut the network ports down, while the Aastra will. On another note, in general the Polycoms give me less problems. The Aastras are not yet that stable. See my next post to the list. On 1/22/07, Bruce Reeves [EMAIL PROTECTED] wrote: Your list seems to lean heavily to the Aastra, while I choose the Polycom 501/601 over the Aastra, I did like the unit I tested and the cordless. In the end the fact that most of the people using the phones would use the speaker phone, Polycom and their history of conference phones made the choice. We rolled 75 phones at one site and another 30 now at remote locations. As far a a receptionist phone, we choose to use a software operator panel instead of a phone that took up most of the desk, there were initial concerns but the results have been excellent. If you have not already done so grab a few people from different parts of the office and have them give their 2 cents, it will help to have their perspectives on the quality and feel of the phones. On 1/22/07, Vikas [EMAIL PROTECTED] wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --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 -- Bruce Nortex Networks ___ --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 ___ --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 ### This message has been scanned by
Re: [asterisk-users] Asterisk 1.4 Polycom buddy status
We also use Polycom IP650 phones. They are assigned to our customer service department. Each SIP interface is a member of our customer service Queue in Asterisk. The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. When this happens there is no SIP channel and the SIP peer appears normal. I have been unable to isolate a procedure to duplicate the problem. It happens erratically to all member interfaces throughout the day. I know that removing the call-limit option from the device's config will stop the problem. This will also remove the ability for the SIP channel driver to track the device's state so we can't remove it permanently. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. James Andrewartha wrote: Olle E Johansson wrote: 23 jan 2007 kl. 16.09 skrev Chris Bullock: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. I've seen reports on it, but haven't been able to repeat this. I need to know a way to force this to happen, repeatably. If I can get that, I can propably trace it and fix it. It can also happen if you have packet loss in the network, of course. I've seen it happen when asterisk restarts (or possibly even just reloads SIP) without the phone being restarted - it's generally accompanied by -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.0.51 on the console. I think the status gets stuck as available most of the time, but you don't notice it because that's the default. ___ --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 1.4 Polycom buddy status
Our 650s are running 2.0.3b. The problem still exists for us. We see the devices as members of our customer service queue stick on 'in-use' in the Queue application while the device has no active SIP channel and will accept calls. Removing 'call-limit' from the sip.conf in Asterisk for the device will fix the issue. This however will also keep the SIP channel driver in Asterisk from tracking the state of the device. Bryan M. Johns wrote: I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Thanks, Bryan M. Johns Partner *Shelton | Johns Technology Group* office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 *http://www.sheltonjohns.com* http://www.sheltonjohns.com/ On Jan 23, 2007, at 10:09 AM, Chris Bullock wrote: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --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 ___ --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 prun realtime peer only can't prune user
Hi All, I'm running 1.2.9.1. I can prune sip realtime peers and users and iax realtime peers but no command to prune iax realtime users. Was this implemented in a later version? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] iax.conf setvar= like sip.conf setvar=?
Hi All, I'm running 1.2.9.1, is setvar= implemented in iax.conf in a later version of asterisk? If so, which one? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Agent Pre Acknowledgement Message
I'm trying to wean my self using the Agent channel in Asterisk. The main
reason I use it is for the callback acknowledgement , where the user
presses # to finally acknowledge the call. I have implemented this in the
Dialplan using the Macro in the docs. This works well as long as the user
enters something.
However if the user hangs up , then both sides of the bridge call are hung
up as well , i.e. the customer calling waiting in the Queue, is hungup on.
I believe the problem is because the Read section of the dial plan will
return a -1 when the operator being called hangs up on it.
Has anyone else spotted this ? Anyone Solved the problem ?
[macro-screen]
exten=s,1,Wait(.25)
exten=s,2,Read(ACCEPT|screen-callee-options|1)
exten=s,3,Gotoif($[${ACCEPT} = 1] ?50)
exten=s,4,Gotoif($[${ACCEPT} = 2] ?30)
exten=s,5,Gotoif($[${ACCEPT} = 3] ?40)
exten=s,6,Gotoif($[${ACCEPT} = 4] ?30:30)
exten=s,30,Set(MACRO_RESULT=CONTINUE)
exten=s,40,Read(TEXTEN|custom/screen-exten|)
exten=s,41,Gotoif($[${LEN(${TEXTEN})} = 3]?42:45)
exten=s,42,Set(MACRO_RESULT=GOTO:from-internal^${TEXTEN}^1)
exten=s,45,Gotoif($[${TEXTEN} = 0] ?46:4)
exten=s,46,Set(MACRO_RESULT=CONTINUE)
exten=s,50,Playback(after-the-tone)
exten=s,51,Playback(connected)
exten=s,52,Playback(beep)
Here is the console log for the even,
-- SIP/cosip-peer-09bfd798 answered Zap/25-1
-- Zap/1-1 answered Local/[EMAIL PROTECTED],2
-- Executing Wait(Zap/1-1, .25) in new stack
-- Executing Read(Zap/1-1, ACCEPT|custom/this-is-helpdesk|1) in
new stack
-- Accepting a maximum of 1 digits.
-- Playing 'custom/this-is-helpdesk' (language 'en')
-- Hungup 'Zap/25-1'
Destroying call '[EMAIL PROTECTED]'
-- Channel 0/1, span 1 got hangup request
-- User disconnected
Jan 24 15:13:12 WARNING[31943]: res_features.c:1384 ast_bridge_call:
Bridge failed on channels Local/[EMAIL PROTECTED],2 and Zap/1-1
-- Hungup 'Zap/1-1'
-- Agent/100 answered Zap/26-1
-- Stopped music on hold on Zap/26-1
-- Hungup 'Zap/26-1'
Thanks,
Diarmaid.
___
--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] ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
On Wed, 24 Jan 2007 09:11:20 +
Gavin Hamill [EMAIL PROTECTED] wrote:
Processing does not continue to the NoOp or Dial - what am I doing
wrong? I've also tried with the 'j' option to 'jump to priority n+101'
even though I'm using AEL, but it's made no difference.
For the benefit of the archive
I got this working by using a 'catch h {...}' block
at the bottom of the macro rather than switch'ing on the variables
set by ChanIsAvail().
Cheers,
Gavin.
___
--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] Getting confused on signalling mode Vs framing and encoding: T1 CAS
Hello, I'm trying to make my asterisk box to act as a telco, in order to reproduce a US environment in europe. Our telco provider is giving us those settings: ESF B8ZF Inbound = EM Immediate Outbound sig =Wink Start Yield to Glare = Yes Those trunks are using CAS for signaling. I have tried many configs/combinations in zaptel.conf and zapata-channels.conf In zaptel.conf, when having something like span=5,0,0,cas,b8zs and in zapata-channels something like signalling=featb I end-up in the log file with something like chan_zap.c: Got hook complete in MF FGD, waiting for wink now on channel 125 If in zaptel.conf I put something like span=5,0,0,esf,b8zs My call is immediatly stopped: Jan 24 17:17:33 DEBUG[25076] chan_zap.c: Got event On hook(1) on channel 125 (index 0) Jan 24 17:17:33 DEBUG[25076] channel.c: Didn't get a frame from channel: Zap/125-1 Jan 24 17:17:33 DEBUG[25076] channel.c: Bridge stops bridging channels SIP/2707-0083be10 and Zap/125-1 What is the correct zaptel.conf for my case ? If I specify esf in zaptel.conf, where do I mention I do CAS for RBS ? Thanks. V.Thinselin ___ --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] Incoming SIP line does not display CallerID correctly
Lee Jenkins wrote:
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box, but then figured out that they are sending the call to an
extension that matches my number with them, in the context defined in
sip.conf. So instead of transferring the call to say, incoming they
are sending the call to incoming/55 where 55 is the
sip phone number I have with them.
So, I have to account for that extension with something like this:
[incoming]
exten=55,Goto(incoming,s,1)
Thus transferring the call to the context that I want it to come in on.
The problem that I have is the caller ID ${CALLERID(num)} always shows
the actual number provided by Telasip and not the actual caller id
information.
I also have axVoice and they do not do it this way. They simply send it
to the context without specifying an extension.
Below is a sip packet. The Caller ID comes through correctly on the sip
packet by for some reason as I mentioned, Asterisk is reporting it as
the number I have with the sip provider.
Below is the sip packet. The 33 represents my cell phone I
was using to call into the system, which was correct.
localhost*CLI exit
-- SIP read from 4.79.19.56:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:4.79.19.56;ftag=as5bfea671;lr=on
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0
Via: SIP/2.0/UDP
4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK3f2e414e;rport=5060
From: 33 sip:[EMAIL PROTECTED];tag=as5bfea671
To: sip:[EMAIL PROTECTED];tag=as12b47a8d
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: 33
sip:[EMAIL PROTECTED];privacy=off;screen=no
Content-Length: 0
P-hint: proxy loose route
Just a bit more information on this in hopes that someone more
knowledgeable than I will chime in.
This problem still persists, of course and I'm having a pickle trying to
figure out what the problem would be.
Summary:
I have two SIP accounts. One with axVoice who forwards the call to my
asterisk box at the s extension of my incoming context. The other
with Telasip who forwards the calls to 55 extension of my
incoming context where 55 represents the provisioned telephone
number for that line.
The problem is that with calls coming in from the Telasip account,
Asterisk ${CALLERID(num)} always returns 55, the Telasip
provisioned phone number instead of the caller's real CID.
My incoming extension.conf:
; -- Before adding Telasip account
[incoming]
exten=s,1,Answer()
exten=s,2,Ringing()
exten=s,3,SetMusicOnHold(default)
exten=s,4,Wait(1)
exten=s,5,Goto(check_time,s,1)
... check_time does a quick GoToIfTime and into the dialplan we go.
When I got my Telasip account, I had to change it to match the extension
they were pointing to
[incoming]
exten=s,1,Answer()
exten=s,2,Ringing()
exten=s,3,SetMusicOnHold(default)
exten=s,4,Wait(5)
exten=s,5,Goto(check_time,s,1)
exten=55,1,Answer()
exten=55,2,Ringing()
exten=55,3,SetMusicOnHold(default)
exten=55,4,Wait(5)
exten=55,5,Goto(check_time,s,1)
; exten=55,1,Goto(incoming,s,1 = Tried this too.
I've tried it a couple of different ways and while the calls do get
routed correctly and everything works fine otherwise, any reference to
${CALLERID(num)} returns the SIP number called and not the caller's
correct CID. For instance, inserting a Noop() such as ;
exten=55,2,Noop(caller id: ${CALLERID(num)})
results in caller id: 55 in the CLI when the call comes in.
Notice from the priority that I tried it after a call to Answer().
I'm a little stumped so any suggestions or observations would be
appreciated as always.
--
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] Re: Disconnect Supervision UK / BT solution?
Sure -- this is for my Sangoma a200 -=-=-=- Zapata.conf (currently supporting CallerID) [channels] echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 musiconhold=default callwaiting=no transfer=yes usecallingpres=yes cadence=200,200,200,4000 usecallerid=yes callerid=asreceived cidsignalling=v23 cidstart=polarity sendcalleridafter=0 ;configure incoming lines group=1 context=UkExt rxgain=6 txgain=1 signalling=fxs_ks channel = 1-2,5-6 ;configure outgoing lines group=2 callgroup=2 pickupgroup=2 context=UkInt rxgain=-1 txgain=2 signalling=fxo_ks channel = 3-4,7-8 -=-=-=-=-=-=- -=-=-= Zaptel.conf =-=-=- fxsks=1-2,5-6 fxoks=3-4,7-8 loadzone = uk defaultzone=uk Ideas? -- Chris Carlos Rojas [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hello, What's your zapata.conf and zaptel.conf? On 1/20/07, Matt Brown [EMAIL PROTECTED] wrote: Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content 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] Getting confused on signalling mode Vs framing and encoding: T1 CAS
On Jan 24, 2007, at 10:20 AM, Thinselin, Vincent wrote: Hello, I'm trying to make my asterisk box to act as a telco, in order to reproduce a US environment in europe. Our telco provider is giving us those settings: ESF B8ZF Inbound = EM Immediate Outbound sig =Wink Start Yield to Glare = Yes Those trunks are using CAS for signaling. I have tried many configs/combinations in zaptel.conf and zapata- channels.conf In zaptel.conf, when having something like span=5,0,0,cas,b8zs and in zapata-channels something like signalling=featb try em_w: E M Wink Start I end-up in the log file with something like chan_zap.c: Got hook complete in MF FGD, waiting for wink now on channel 125 If in zaptel.conf I put something like span=5,0,0,esf,b8zs My call is immediatly stopped: Jan 24 17:17:33 DEBUG[25076] chan_zap.c: Got event On hook(1) on channel 125 (index 0) Jan 24 17:17:33 DEBUG[25076] channel.c: Didn't get a frame from channel: Zap/125-1 Jan 24 17:17:33 DEBUG[25076] channel.c: Bridge stops bridging channels SIP/2707-0083be10 and Zap/125-1 What is the correct zaptel.conf for my case ? If I specify esf in zaptel.conf, where do I mention I do CAS for RBS ? you dont - signalling type sets it Thanks. V.Thinselin ___ --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 1.4 Polycom buddy status
James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. ___ --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] Panasonic Hybrid Integration Advice Needed
If you use a Vegastream gateway on the actual incoming ISDN circuits then you won't even need to touch the Panasonic to integrate both systems. Regards Scott Pinhorne VoxIT Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: 24 January 2007 15:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Panasonic Hybrid Integration Advice Needed Which panasonic system? I'm assuming you are talking about the TDA line. If so get a IP Gateway card on the TDA system, that card uses h323, then configure it with asterisk as h323, or my favorite, get a PRI card on the TDA sysem (unless it's a TDA50 then the option is not available), and a T1 card on asterisk, and create a dialplan on the Panasonic that goes out over the PRI card. On 1/24/07, John French [EMAIL PROTECTED] wrote: I have a client who has a Panasonic Hybrid system. They are taking in another company as a building tenant and the tenant will be on a new 12 station Asterisk system. This new asterisk system will have 4 FXO ports plus ITSP. The two systems will be separate except that they should tie together for the purposes of dialing extensions directly on the opposite phone system and for transferring calls. I'm looking for advice on how best to accomodate this. Is it possible to do this via the Panasonic's IP interface or will I need to cross connect them via T1 cards? This is my first integration as you can probably surmise. Thanks in advance. -- 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 ___ --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] Queues Question
Michiel van Baak wrote: On 17:36, Thu 18 Jan 07, Lee Jenkins wrote: Michiel van Baak wrote: exten=999,1,Queue(support,tr|||60) and never put it back. From there it was a downward spiral ;) If you remove the r, does that fix the issue ? Yes. It did. Still not sure why it didn't work originally without the r flag, prompting me to try it with the r flag. -- 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] iax2 prun realtime peer only can't prune user
On 1/24/07, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I'm running 1.2.9.1. I can prune sip realtime peers and users and iax realtime peers but no command to prune iax realtime users. Was this implemented in a later version? Thanks. JR From what I could dig up, it looks like you can do peers or all, but not users. The code in iax2.c has a function prune_users(); but I could not find anything pointing to a CLI command to prune iax users. Regards, 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] OT - Cisco 7960 functionality
Can anyone point me to info on how to change the functionality of the SIP (7.4) 7960's. We previously had an SCCP firmware on the phone and the users want the phone to work like it used too. Here are some examples: The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want to push the answer softkey after they pickup the handset in order to answer a call. The users want the transfer softkey on the screen while on a call. Currently it is acessable via the more softkey. Any help or suggestions would be appreciated. I have been looking but have only found docs on how to use the phone features as they are and how to use external programs through xml. Thanks, Andy ___ --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 1.4 Polycom buddy status
Yeah, we don't use Buddy Watch. We don't use call-limit because we want call-limit. We use it because it's the only way, that I'm aware of, to get the SIP channel driver to monitor the state of the member SIP interface. We use autopause=yes and ringinuse=no in our customer service queue configuration. Without specifying call-limit, the Queue application continues to send new calls to member interfaces that are in-use or busy. The SIP device replies to Asterisk saying it's busy and the Queue application pauses the member because of autopause. With call-limit enabled and set to any number, the Queue application knows that the member interface is busy and will not send new calls. I replied to this post describing our findings with the Queue application because it sounds like the same behavior occurs with hints and buddy watch. The state detection in the SIP channel driver appears suspect to me. Eric ManxPower Wieling wrote: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. ___ --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] convert URI string to lowercase
any idea, how to do something like this, but in correct/functional form?
;-)
Set(foo=System(echo ${EXTEN} | tr [:upper:] [:lower:]))
${EXTEN} is SomeStrinG
${foo} output should bee somestring
___
--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] Semi OT - Point to Point FXO/FXS Gateway Communication
Has anyone had any experience using FXO and FXS gateways to extend legacy PBX extensions to remote users? I have a customer who needs to do this, but wants seamless, two way communication, with a SIP server and without the need for 2-stage dialing. If anyone has any experience with a solution please let me know. Thanks Cory Andrews ___ --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] Getting confused on signalling mode Vs framingand encoding: T1 CAS
ESF B8ZF Inbound = EM Immediate Outbound sig =Wink Start Yield to Glare = Yes In zaptel.conf, when having something like span=5,0,0,cas,b8zs and in zapata-channels something like signalling=featb try em_w: E M Wink Start Jerry is right - you need to set signaling in zaptel.conf like this... signalling=em_w ... so that it matches what's in zapata.conf. 'featb' is a reference to 'Feature Group B' which is for ISDN and not for good ole CAS/RBS. -MC ___ --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] OT - Cisco 7960 functionality
The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want to push the answer softkey after they pickup the handset in order to answer a call. Doesn;t it answers when you pick up the handset? Here it does so... The users want the transfer softkey on the screen while on a call. Currently it is acessable via the more softkey. I've asked Cisco whether all the above can be done and got a negative reply. __Yehavi: ___ --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] vxml support
Can Asterisk support vxml? Can i work with Asterisk and vxml? Is there any AGI framework that can use vxml? It seems like support is still a bit limited, but evidently it is available: http://www.voip-info.org/wiki/view/VoiceXML HtH, MC ___ --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] channel name
Hello everybody, I was wondering if anybody knows how to make channel IDs different if all call are coming from the same host: core show channels Channel Location State Application(Data) SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback() SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback() Thanks in advance. Serge ___ --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] OT - Cisco 7960 functionality
another reason, why better is to completely avoid ci$co phones when used with anything other than callmanager ;-) Yehavi Bourvine +972-8-9489444 wrote: The users want the transfer softkey on the screen while on a call. Currently it is acessable via the more softkey. I've asked Cisco whether all the above can be done and got a negative reply. __Yehavi: ___ --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] setting up AMD
I'm trying get this working. I've looked through the list, and can't see
how to get AMD to print out more. I have it call and say Hello like I
normally would. I've tried to say more and less doesn't seem to matter.
After I hangup it does recognize hangup. Here's logging during an attempt
where I make outbound call and answer, but then hangup after 1-2 seconds:
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence [8000]
greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000]
minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5]
silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call '
[EMAIL PROTECTED]'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
The amd.conf:
[amd]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
___
--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] Dell Server Question
Hi all, We're planning an Asterisk implementation consisting of two SC1425 Dell Servers using Digium T1 cards and SATA drives. The problem I'm having is the only PCI slot shares an IRQ with the SATA controller. Any altering of one device's IRQ takes the other device's IRQ with it in lockstep. I've disabled all non-essential integrated devices with no change. I'm a little worried about call quality if I can't resolve this IRQ sharing issue. So I'm wondering a few things... 1. Am I missing something with regard to the IRQ configuration? 2. Should I trust in the motherboard's APIC and quit worrying? 3. Assuming there is no solution to (1), is anyone else running a similiar setup with no call quality concerns? Thanks! Nick ___ --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] setting up AMD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't
see how to get AMD to print out more. I have it call and say Hello like
I normally would. I've tried to say more and less doesn't seem to
matter. After I hangup it does recognize hangup. Here's logging during
an attempt where I make outbound call and answer, but then hangup after
1-2 seconds:
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
[8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
[5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
The amd.conf:
[amd]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
Peter,
It looks like your initial silence setting might be having trouble. The
amd.conf file has a value of 3500 but the log file is showing 8000. Try
changing the amd.conf to something like 3000 and issue a reload at the
CLI. Make another test call and see if the trace still shows 8000 for
the initial silence. I think having an initial silence value that is
longer than the total analysis time might be causing the undesired
behavior.
Let us know what happens when you try to modify the initial silence
value.
-MC
___
--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 parking causes Asterisk to crash
I have one system that is crashing everytime a call is parked and I have tried recompiling the asterisk, checking out the latest SVN of 1.2 and modifying the configuration. I have identified what I think is the error and have back traces but since this is occurring on only one system I want to know what might cause this. CLI: -- SIP/xlite_brr-098d1e98 is ringing -- SIP/xlite_brr-098d1e98 answered IAX2/192.168.0.231:4569-1 -- Started music on hold, class 'default', on IAX2/192.168.0.231:4569-1 -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on IAX2/192.168.0.231:4569-1 -- Started music on hold, class 'default', on IAX2/192.168.0.231:4569-1 == Parked IAX2/192.168.0.231:4569-1 on 701. Will timeout back to extension [inside] 1513, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') == Auto fallthrough, channel 'IAX2/192.168.0.231:4569-1' status is 'ANSWER' -- Stopped music on hold on IAX2/192.168.0.231:4569-1 -- Hungup 'IAX2/192.168.0.231:4569-1' == IAX2/192.168.0.231:4569-1 got tired of being parked -- Hungup 'IAX2/192.168.0.231:4569-1' Jan 24 13:43:26 WARNING[24727]: channel.c:897 ast_channel_free: Unable to find channel in list pbx*CLI Disconnected from Asterisk server The back trace has a similar message about channel.c #6 0x080616bd in ast_channel_free (chan=0x9932c48) at channel.c:864 cur = Variable cur is not available. Has anyone run into this before? I cannot find any difference between this system and the others I have deployed with the same hardware and configurations. -- Bruce Nortex Networks ___ --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] Disconnected Calls
Hello. I am running asterisk 1.2.14 on a Dell poweredge with a Digium FXO/FXS card connected to 6 analog lines and using Linksys spa942 phones. My users are complaining of randomly disconnected calls, and when I watch the log (debug warning,notice,error), I don't see any cause. It looks like asterisk is seeing a hangup from the analog end. I have attached my zaptel.conf and zapata.conf. What additional information can I provide to make this an intelligent question? Many Thanks, Ejay Hire Zapata.conf ; Zapata telephony interface [trunkgroups] [channels] musiconhold=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=1200 busydetect=yes callgroup=1 pickupgroup=1 immediate=no group=0 language=en context=default rxgain=12.4 txgain=4 signalling=fxs_ls rxwink=300 ; Atlas seems to use long (250ms) winks relaxdtmf=yes channel = 1 channel = 2 channel = 3 channel = 4 channel = 5 channel = 7 channel = 8 group=1 channel = 6 Zaptel.conf cat /etc/zaptel.conf # Span 1: WCTDM/0 Wildcard TDM2400P Board 1 fxsls=1 fxsls=2 fxsls=3 fxsls=4 fxsls=5 fxsls=6 fxsls=7 fxsls=8 loadzone= us defaultzone = us ___ --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] setting up AMD
now I have amd.conf set to this:
initial_silence = 3700
greeting = 2500
after_greeting_silence = 1200
total_analysis_time = 6000
min_word_length = 100
between_words_silence = 50
maximum_number_of_words = 4
silence_threshold = 860
The resulting log is this:
Jan 24 18:53:04 DEBUG[31555] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]
Jan 24 18:53:04 VERBOSE[31567] logger.c: -- AMD:
SIP/sip.broadvoice.com-087743a0 6079362172 (null) (Fmt: 4)
Jan 24 18:53:04 VERBOSE[31567] logger.c: -- AMD: initialSilence [3700]
greeting [2500] afterGreetingSilence [1200] totalAnalysisTime [6000]
minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4]
silenceThreshold [860]
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Scheduled a registration timeout
for sip.broadvoice.com id #19
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 104: Match
Found
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Registration successful
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Cancelling timeout 19
Jan 24 18:53:28 DEBUG[31555] chan_sip.c: Auto destroying call '
[EMAIL PROTECTED]'
Jan 24 18:53:32 DEBUG[31555] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 102: Match Found
Jan 24 18:53:36 VERBOSE[31567] logger.c: -- AMD: HANGUP
Jan 24 18:53:36 DEBUG[31567] app_amd.c: Got hangup
Jan 24 18:53:36 DEBUG[31567] pbx.c: Extension s, priority 2 returned
normally even though call was hung up
Jan 24 18:53:36 DEBUG[31567] chan_sip.c: update_call_counter(6079362172) -
decrement call limit counter
On 1/24/07, Michael Collins [EMAIL PROTECTED] wrote:
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Peter Halliday
*Sent:* Wednesday, January 24, 2007 11:56 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't see
how to get AMD to print out more. I have it call and say Hello like I
normally would. I've tried to say more and less doesn't seem to matter.
After I hangup it does recognize hangup. Here's logging during an attempt
where I make outbound call and answer, but then hangup after 1-2 seconds:
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence [8000]
greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000]
minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5]
silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call '
[EMAIL PROTECTED]'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
The amd.conf:
[amd]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
Peter,
It looks like your initial silence setting might be having trouble. The
amd.conf file has a value of 3500 but the log file is showing 8000. Try
changing the amd.conf to something like 3000 and issue a reload at the
CLI. Make another test call and see if the trace still shows 8000 for the
initial silence. I think having an initial silence value that is longer
than the total analysis time might be causing the undesired behavior.
Let us know what happens when you try to modify the initial silence
value.
-MC
___
--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] Semi OT - Point to Point FXO/FXS Gateway Communication
Cory, it's called dialplan magic it realy depends what PBX it is, not all of them allow dial plan magic. But it is possible on most pbxes. On 1/24/07, Cory Andrews [EMAIL PROTECTED] wrote: Has anyone had any experience using FXO and FXS gateways to extend legacy PBX extensions to remote users? I have a customer who needs to do this, but wants seamless, two way communication, with a SIP server and without the need for 2-stage dialing. If anyone has any experience with a solution please let me know. Thanks Cory Andrews ___ --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] Panasonic Hybrid Integration Advice Needed
I disagree on this, you will have to create a dialplan in the panasonic to tell it when to go over the ISDN circuit. On 1/24/07, Scott Pinhorne [EMAIL PROTECTED] wrote: If you use a Vegastream gateway on the actual incoming ISDN circuits then you won't even need to touch the Panasonic to integrate both systems. Regards Scott Pinhorne VoxIT Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: 24 January 2007 15:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Panasonic Hybrid Integration Advice Needed Which panasonic system? I'm assuming you are talking about the TDA line. If so get a IP Gateway card on the TDA system, that card uses h323, then configure it with asterisk as h323, or my favorite, get a PRI card on the TDA sysem (unless it's a TDA50 then the option is not available), and a T1 card on asterisk, and create a dialplan on the Panasonic that goes out over the PRI card. On 1/24/07, John French [EMAIL PROTECTED] wrote: I have a client who has a Panasonic Hybrid system. They are taking in another company as a building tenant and the tenant will be on a new 12 station Asterisk system. This new asterisk system will have 4 FXO ports plus ITSP. The two systems will be separate except that they should tie together for the purposes of dialing extensions directly on the opposite phone system and for transferring calls. I'm looking for advice on how best to accomodate this. Is it possible to do this via the Panasonic's IP interface or will I need to cross connect them via T1 cards? This is my first integration as you can probably surmise. Thanks in advance. -- 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 ___ --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 ___ --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] realtimeinsert and realtimedelete functions
Hi, In the system, there are realtime and realtimeupdate to access data in realtime model. Does it include realtimeinsert and realtimedelete such that they can be used to manipulate the database more completely? ___ --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] Best way to connect analog modem
Hello Asterisk fans, I try to connect an analog modem to Asterisk. The modems are connected e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm using a Wildcard TE110P (E1). Is it possible to connect the modems to an ATA? Which ATA I should use for that scenarios? Cheers Bastian Virus checked by G DATA AntiVirusKit Version: AVK 17.1806 from 04.01.2007 Virus news: www.antiviruslab.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] Re: Grandstream GXP2000 and Interception of call ?
Noc Phibee wrote: Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone It's called Call Pickup. http://www.voip-info.org/wiki-PBX+Call+Pickup ___ --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] Multiple parking lot
Hi list, Does anyone know any ways to have mutiple parking lots? I've got a pbx that 2 customers share, both need their own, and then have lights on the phone flash when they park the call (snom phones). Any ideals I'm not thinking of?!? Any help would be great! Thanks Ron ___ --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
Polycom Firmware -- Was: [asterisk-users] Asterisk 1.4 Polycom buddy status
I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth ___ --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] setting up AMD
Hmm... not too sure what's up with this one. I've only used AMD with
Zap channels, so I don't know if there are any hidden gotchas with using
SIP.
Has anyone else used app_amd with SIP calls?
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up AMD
now I have amd.conf set to this:
initial_silence = 3700
greeting = 2500
after_greeting_silence = 1200
total_analysis_time = 6000
min_word_length = 100
between_words_silence = 50
maximum_number_of_words = 4
silence_threshold = 860
The resulting log is this:
Jan 24 18:53:04 DEBUG[31555] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]
Jan 24 18:53:04 VERBOSE[31567] logger.c: -- AMD:
SIP/sip.broadvoice.com-087743a0 6079362172 (null) (Fmt: 4)
Jan 24 18:53:04 VERBOSE[31567] logger.c: -- AMD: initialSilence
[3700] greeting [2500] afterGreetingSilence [1200] totalAnalysisTime
[6000] minimumWordLength [100] betweenWordsSilence [50]
maximumNumberOfWords [4] silenceThreshold [860]
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Scheduled a registration
timeout for sip.broadvoice.com id #19
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 104:
Match Found
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Registration successful
Jan 24 18:53:20 DEBUG[31555] chan_sip.c: Cancelling timeout 19
Jan 24 18:53:28 DEBUG[31555] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED] '
Jan 24 18:53:32 DEBUG[31555] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jan 24 18:53:36 VERBOSE[31567] logger.c: -- AMD: HANGUP
Jan 24 18:53:36 DEBUG[31567] app_amd.c: Got hangup
Jan 24 18:53:36 DEBUG[31567] pbx.c: Extension s, priority 2 returned
normally even though call was hung up
Jan 24 18:53:36 DEBUG[31567] chan_sip.c: update_call_counter(6079362172)
- decrement call limit counter
On 1/24/07, Michael Collins mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't
see how to get AMD to print out more. I have it call and say Hello like
I normally would. I've tried to say more and less doesn't seem to
matter. After I hangup it does recognize hangup. Here's logging during
an attempt where I make outbound call and answer, but then hangup after
1-2 seconds:
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
[8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
[5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] '
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
The amd.conf:
[amd]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
Peter,
It looks like your initial silence setting might be having trouble. The
amd.conf file has a value of 3500 but the log file is showing 8000. Try
changing the amd.conf to something like 3000 and issue a reload at the
CLI. Make another test call and see if the trace still shows 8000 for
the initial silence. I think having an initial silence value that is
longer than the total analysis time might be causing the undesired
behavior.
Let us know what happens when you try to modify the initial silence
value.
-MC
___
--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
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and
[asterisk-users] SPA3K to SPA3K DTMF issue
Hi all, Has anyone faced an issue when sending DTMF from FXS of one SPA3K to FXO of another SPA3K through asterisk? Im not able to send it properly. Wanna be sure if its an issue faced by all.. If you have a fix for it, pls guide me. Thanks Dan ___ --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] NAT solutions
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote: In the meanwhile, use IAX, which understands about NAT pretty well. If you have multiple SIP phones on a LAN behind a NATing router, just put a small asterisk box on the LAN. It can manage your hairpin calls internally, save you bandwidth by trunking the IAX traffic to the central asterisk and avoid all the NAT hassle by using a single port (outgoing) and refreshing it often enough for the router to hold it open. Tim Panton www.mexuar.net www.westhawk.co.uk/ IAX is a fine protocol as far as it goes, however this answer is really not a workable one. There are only a few IAX phones, and they are not nearly as solid and full featured as the many SIP phones. There are some IAX termination and origination providers, but there are far more SIP providers. For a remote phone, not on the same network as the Asterisk box (in which event the NAT worries are different) you definitely want to use the same protocol for the phone as for your term/orig provider. Otherwise you will be forced to hairpin your audio through your asterisk server, adding latency and wasting bandwidth and cpu for little reason. In addition, many people just want to do things like give family or employees a phone they can take home, or take to a remote location and use on the PBX. They probably can't just put up an Asterisk server to make this happen, and nor should they want to. An additional server is not only more work and requires an always-on server computer, it's another thing that can go wrong. No thanks. Even if you can run Asterisk on a WRT54G, and thus don't have the $200/year power expense of a server, it's still not what you really want. IAX is great but SIP is also a reality, and putting Asterisk into the just works category is a really important milestone. One I think that is intended to be improved a lot for 1.6. ___ --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] OT: High Quality Wireless Headset for Cisco IP Phones and *
On Tue, Jan 23, 2007 at 02:01:43PM -0600, Tom wrote: Has anyone found a high quality wireless headset that works well with Cisco 7960 IP phones on an asterisk system? I tried the vxxi offering but the sound quality was pretty bad. Since these are pricey, I don't want to sample blindly. I've got one of the Plantronics bluetooth ones. It's OK, but frankly, with bluetooth hardware costing just a couple of bucks, you would think we should just see bluetooth becoming standard in every non-budget IP phone. People already have the headsets in many cases, and you can go digital all the way, and even rely on the headset's echo cancellation if you like. SNOM has a high end phone with this but otherwise it's been much slower to come than you would think. Alas, this doesn't really answer your question. ___ --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] issue with ivtv wctdm zaptel drivers (TDM PCI Master abort)
Hi,
I'm experiencing an issue with my x86_64 machine containing a Hauppauge
PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards.
Independently of each other both cards work fine, but once the wctdm driver
is loaded and mythtv tries to record something on the PVR-500 the wctdm
driver freaks out. The error message is see is TDM PCI Master abort
printed over and over again in the syslog.
I don't know if this is the ivtv or wctdm drivers fault, but the issue only
shows up between these drivers. Burning a dvd works, usb devices work, etc.
I've tried the following:
- Moving the TDM400p to a PCI slot so its not sharing an interrupt
with an other device
- ivtv drivers: 0.9.1 and the 0.10.0
- zaptel 1.4.0 (tagged) and the latest from the 1.4 branch (svn: 1952)
- I don't recall having this error when the machine was still running
in 32bit mode (i586 kernel and i586 userland), right now the machine is
running with a x86_64 kernel and x86_64 userland.
This seems to be the part in
http://svn.digium.com/view/zaptel/tags/1.4.0/wctdm.c that prints those error
messages to the syslog:
if (ints 0x10) {
/* Stop DMA, wait for watchdog */
printk(TDM PCI Master abort\n);
wctdm_stop_dma(wc);
And this is what I'm seeing in syslog:
Jan 24 20:04:06 taz kernel: ivtv0 irq: ENC START CAP 0: 0016d6c0 4000
Jan 24 20:04:06 taz kernel: ivtv0 dma: DMA encoder MPEG: 0x4000 bytes at
0x0016d6c0
Jan 24 20:04:06 taz kernel: ivtv0 dma: start DMA for encoder MPEG
Jan 24 20:04:06 taz kernel: ivtv0 irq: ENC DMA COMPLETE 3 0
Jan 24 20:04:06 taz kernel: ivtv0 dma: DMA encoder MPEG completed (16d6c0)
Jan 24 20:04:06 taz kernel: ivtv0 ioctl: read 4096 bytes from encoder MPEG
Jan 24 20:04:06 taz kernel: ivtv0 irq: ENC START CAP 0: 001716c0 6800
Jan 24 20:04:06 taz kernel: ivtv0 dma: DMA encoder MPEG: 0x6800 bytes at
0x001716c0
Jan 24 20:04:06 taz kernel: ivtv0 dma: start DMA for encoder MPEG
Jan 24 20:04:06 taz kernel: TDM PCI Master abort
Jan 24 20:04:06 taz last message repeated 77 times
Jan 24 20:04:06 taz kernel: ivtv0 irq: ENC DMA COMPLETE 3 0
Jan 24 20:04:06 taz kernel: ivtv0 dma: DMA encoder MPEG completed (1716c0)
Jan 24 20:04:06 taz kernel: TDM PCI Master abort
Jan 24 20:04:06 taz last message repeated 179 times
Jan 24 20:04:06 taz kernel: ivtv0 info: read 4096 from encoder MPEG, got
4096
Jan 24 20:04:06 taz kernel: TDM PCI Master abort
(ivtv 0.10 trunk, debug level = 511).
Could this have something todo with these two pieces of hardware fighting
about DMA?
with kind regards,
Stefan van der Eijk
___
--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] NAT solutions
From: Brad Templeton [EMAIL PROTECTED] On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote: In the meanwhile, use IAX, which understands about NAT pretty well. If you have multiple SIP phones on a LAN behind a NATing router, just put a small asterisk box on the LAN. It can manage your hairpin calls internally, save you bandwidth by trunking the IAX traffic to the central asterisk and avoid all the NAT hassle by using a single port (outgoing) and refreshing it often enough for the router to hold it open. Tim Panton www.mexuar.net www.westhawk.co.uk/ IAX is a fine protocol as far as it goes, however this answer is really not a workable one. There are only a few IAX phones, and they are not nearly as solid and full featured as the many SIP phones. There are some IAX termination and origination providers, but there are far more SIP providers. ... IAX is great but SIP is also a reality, and putting Asterisk into the just works category is a really important milestone. One I think that is intended to be improved a lot for 1.6. I have a really dumb question. It appears that Yahoo, MSN, AIM, you name them, they don't have a NAT problem, and some use SIP. I don't think they all stay in voice path, either. What takes? Yuan Liu ___ --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] Panasonic Hybrid Integration Advice Needed
C F wrote: Which panasonic system? I'm assuming you are talking about the TDA line. If so get a IP Gateway card on the TDA system, that card uses h323, then configure it with asterisk as h323, or my favorite, get a PRI card on the TDA sysem (unless it's a TDA50 then the option is not available), and a T1 card on asterisk, and create a dialplan on the Panasonic that goes out over the PRI card. Speaking as one who has been using the 16 channel IP-GW card with a TDA 100 for over a year, you might want to go with the PRI/T1 solution, although I have no experience with it. Although H323 integration with Asterisk has been flawless, I have had no luck with any form of out of band DTMF transport through the IP-GW. The documentation for the card states that it can do it, but does not state what protocol is used and packet captures show no sign of it. If anyone knows anything more, I would like to find out, as the local Panasonic agents have not been much help. Regards, Richard ___ --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] issue with ivtv wctdm zaptel drivers (TDM PCI Master abort)
Hi, On Thu, 2007-01-25 at 08:03 +0100, Stefan van der Eijk wrote: Hi, I'm experiencing an issue with my x86_64 machine containing a Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards. Independently of each other both cards work fine, but once the wctdm driver is loaded and mythtv tries to record something on the PVR-500 the wctdm driver freaks out. The error message is see is TDM PCI Master abort printed over and over again in the syslog. PVR cards are greedy about irq resources. They need a dedicated irq and normally the irq is held for much time. Is not a card that goes with a TDM one, since TDM cards needs a precise irq timining... Imho, you should not run both cards on same box. I had some luck doing that with a DVB-T card... since being digital cards, the amout of data transferred is lower, so can work with a TDM. But was not a hauppauge card. (I know, also the pvr500 is digital, but you have 2 tuners so double data rate and normally mpeg2 data rate in hw encoders is higher that DVB-T data rate) greetings, Matteo ___ --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] Multiple parking lot
There is an SVN branch with this feature: http://svn.digium.com/view/asterisk/team/oej/multiparking/ I had hope this would be a feature added to Asterisk 1.4, but fail to see it on the changelog. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthy Sent: Wednesday, January 24, 2007 21:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple parking lot Hi list, Does anyone know any ways to have mutiple parking lots? I've got a pbx that 2 customers share, both need their own, and then have lights on the phone flash when they park the call (snom phones). Any ideals I'm not thinking of?!? Any help would be great! Thanks Ron ___ --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] TDM2400 Hardware Echo Cancel
I had the exact same problem, removing the hardware echo fix the problem but this is not a solution for a production system. I'm now using another brand of hardware. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Webster, Andrew Envoyé : 23 janvier 2007 14:42 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] TDM2400 Hardware Echo Cancel I have been having the same problems since installing a TDM2400 with hardware echo canceller. The best way to describe the sound is a background crackle or hiss that just can't be filtered out. Increasing the RX gain just makes the problem worse. SIP to SIP calls are flawless. An acquaintance told me the analog line level is too low, but when plugging a regular phone into the line, the signal is plenty loud enough. I am curious if anyone else had similar issues with the TDM2400 card and if they have resolved it. -- Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Sharples Sent: Tuesday, January 16, 2007 09:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDM2400 Hardware Echo Cancel Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --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 ___ --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] channel name
Hello everybody, I was wondering if anybody knows how to make channel IDs different if all call are coming from the same host: core show channels Channel Location State Application(Data) SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback() SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback() Thanks in advance. Serge ___ --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] issue with ivtv wctdm zaptel drivers (TDM PCI Master abort)
On 1/25/07, Matteo Brancaleoni [EMAIL PROTECTED] wrote: Hi, On Thu, 2007-01-25 at 08:03 +0100, Stefan van der Eijk wrote: Hi, I'm experiencing an issue with my x86_64 machine containing a Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards. Independently of each other both cards work fine, but once the wctdm driver is loaded and mythtv tries to record something on the PVR-500 the wctdm driver freaks out. The error message is see is TDM PCI Master abort printed over and over again in the syslog. PVR cards are greedy about irq resources. They need a dedicated irq and normally the irq is held for much time. Interesting, since the TDM card has the dedicated interrupt, and of the ivtv's is sharing an interrupt. # cat /proc/interrupts CPU0 0: 137171395 IO-APIC-edge timer 1: 2259 IO-APIC-edge i8042 8: 0 IO-APIC-edge rtc 9: 0 IO-APIC-fasteoi acpi 12: 4 IO-APIC-edge i8042 14: 63 IO-APIC-edge ide0 15:1272924 IO-APIC-edge ide1 16: 143614215 IO-APIC-fasteoi wctdm 17: 336936 IO-APIC-fasteoi ivtv0 18: 13382861 IO-APIC-fasteoi [EMAIL PROTECTED]::06:00.0, ivtv1 19:125 IO-APIC-fasteoi ohci1394 20: 196998 IO-APIC-fasteoi NVidia CK804 21: 37615 IO-APIC-fasteoi ehci_hcd:usb2 22: 1 IO-APIC-fasteoi ohci_hcd:usb1 23: 18737602 IO-APIC-fasteoi libata, eth1 283:2036011 PCI-MSI-edge eth0 NMI: 4731 LOC: 137148173 ERR: 0 Is not a card that goes with a TDM one, since TDM cards needs a precise irq timining... So, due to the precise timing needs of the TDM card, when the ivtv cards are working, the TDM card doesn't get enough attention and freaks out? Imho, you should not run both cards on same box. Ouch. I'm not happy to hear this :-( I had some luck doing that with a DVB-T card... since being digital cards, the amout of data transferred is lower, so can work with a TDM. But was not a hauppauge card. (I know, also the pvr500 is digital, but you have 2 tuners so double data rate and normally mpeg2 data rate in hw encoders is higher that DVB-T data rate) PVR500 has got 2 analog tuners (in my case: PAL). greetings, Matteo with kind regards, Stefan ___ --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
