Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
2006/4/17, Nicholas Kathmann [EMAIL PROTECTED]: I agree with Lee.I have about 30 machines in production using iaxmodemand hylafax which work perfectly.Most are running off of T1s, but someare on TDM400 and TDM2400s.I only use IBM servers (which are about twice the cost for the low end Dells), and have never had to resolve anIRQ problem.I just looked up the hylafax usage reports on those peoplerunning the analog FXOs, and one of them had 390 pages in the last week, only one error, which I would consider acceptable.Thanks,Nick1. Do you mean Hylafax and Asterisk are installed on the machine and share the same TDM cards ?2. If positive, do you have any extension which is used for both voice and fax ? For instance, user Alice receives voice or fax calls on its own extension. When it's a fax, your server detects it and and let Hylafax get the call.CheersOlivier ___ --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] Quick question
Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! -- Tomislav Parcina tparcina#lama.hr ___ --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 hardware for new office suggestion
Simone wrote: I want to thank you for the suggestions. The office is in the UK, so probably we will go for the ISDN30. I am trying to get a SDSL 2mbit for the line so that bandwidth should not be a problem, the internal LAN will be Gbit as said so the QoS as suggested will be only on the firewall (linux). I have lowered expenses for other equipment so I was thinking of buying a Dell 1800 or 2800 server 2x2,8Ghz 2gb ram to set up Asterisk, know this is a big server but they'll use the ISDN lines and VoIP so virtually there could be 20/25 simultaneous calls. I'll have a look at the wiki and the phones suggested, we'd definitely like phones with internal ethernet switch and PoE capable, I'll try to get an idea of what could work for us. That is way overkill for only asterisk and even 4 T1/E1, if there is no transcoding. If so, I have no idea. Kevin ___ --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] Sip.conf
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? -- Tomislav Parcina tparcina#lama.hr ___ --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 asterisk process ?
On 18 Apr 2006, at 03:20, stevanus wrote: Hmm...my output for getconf GNU_LIBPTHREAD_VERSION is NPTL 2.3.4.. I don't know what it's mean anyway :P And for Lee, I'm configuring my asterisk through amp (now freepbx), but I do some custom configuration manually too ;) I guess Paul is right, I suspect there are bugs in asterisk that haven't been solved like avoiding deadlock on iax problem which I had mentioned before.. Unfortunately, I don't know how to recreate the problem so all I can do if the problem is happened just do some killall - 9 asterisk :(... Regads, Stevanus I'd guess you have a startup script for asterisk that is setting the LD_ASSUME_KERNEL environment variable. To check, find the 'main' asterisk process id (almost always the lowest numbered asterisk process) then look (as root) in the /proc entry, eg: cat /proc/13098/environ | strings | grep LD_ASSUME_KERNEL Tim Panton [EMAIL PROTECTED] ___ --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] Quick question
Tomislav Parčina wrote: Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! The native H.323 driver, chan_h323, does support inband DTMF. Good Luck, Jeremy McNamara ___ --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 asterisk process ?
On Tue, 2006-04-18 at 08:29 +0100, Tim Panton wrote: I'd guess you have a startup script for asterisk that is setting the LD_ASSUME_KERNEL environment variable. To check, find the 'main' asterisk process id (almost always the lowest numbered asterisk process) then look (as root) in the /proc entry, eg: cat /proc/13098/environ | strings | grep LD_ASSUME_KERNEL Now we're getting somewhere. In some old contribs/init.d asterisk scripts there is the following:- # Leave this set unless you know what you are doing. #export LD_ASSUME_KERNEL=2.4.1 While others have nothing or this # Uncomment this ONLY if you know what you are doing. # export LD_ASSUME_KERNEL=2.4.1 -- Dave Cotton [EMAIL PROTECTED] ___ --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 asterisk process ?
Any thoughts as to why only 1 of my boxes has this problem? I'm on a 2.6 kernel so any more ideas? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 18 April 2006 09:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] multiple asterisk process ? On Tue, 2006-04-18 at 08:29 +0100, Tim Panton wrote: I'd guess you have a startup script for asterisk that is setting the LD_ASSUME_KERNEL environment variable. To check, find the 'main' asterisk process id (almost always the lowest numbered asterisk process) then look (as root) in the /proc entry, eg: cat /proc/13098/environ | strings | grep LD_ASSUME_KERNEL Now we're getting somewhere. In some old contribs/init.d asterisk scripts there is the following:- # Leave this set unless you know what you are doing. #export LD_ASSUME_KERNEL=2.4.1 While others have nothing or this # Uncomment this ONLY if you know what you are doing. # export LD_ASSUME_KERNEL=2.4.1 -- Dave Cotton [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] multiple asterisk process ?
On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote: Any thoughts as to why only 1 of my boxes has this problem? Is it really a problem? I'm on a 2.6 kernel so any more ideas? Can someone answer what was the original purpose of the export LD_ASSUME_KERNEL=2.4.1 in the asterisk script? Perhaps Gregory Boehnlein, the author, will be able to tell us. -- Dave Cotton [EMAIL PROTECTED] ___ --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] voicemail use external smtp server for sendingmail
if you need a slim MTA to replace sendmail, in a server that's only acting as an * server and need anything else, you could try nullmailer, a small MTA only capable of smtp via a smarthost. It's so little it sould be considered, at least on embedded-systems. We use it with success... 2006/4/18, Jerry Workman [EMAIL PROTECTED]: Read this, it describes how to set up sendmail to use a SMART_HOST. http://lists.freebsd.org/pipermail/freebsd-questions/2004-February/035328.htmlJerryOn 4/17/06, C F [EMAIL PROTECTED] wrote: Asterisk actualy uses the sendmail as you would in the shell. It is up to sendmail to decided how to process the message, and yes you are right asterisk is *not* the one making the SMTP connection. You can't get asterisk to do it directly since asterisk is not initiating any connections, but submitting it to a program that is, by default that program is sendmail. On 4/17/06, Steve Jones [EMAIL PROTECTED] wrote: I had the same question, and I want to make sure I'm clear.This implies to me that Asterisk itself doesn't use SMTP, but rather dumps a message into some directory that Sendmail on the same box will see and process?I have no problem getting Sendmail to use a smarthost, but am I understanding the Asterisk part of this properly, or is there a way to get Asterisk to DIRECTLY use a smarthost, so that Sendmail doesn't have to be running on the local Asterisk box? Thanks! -Steve -Original Message- From: C F [mailto:[EMAIL PROTECTED]] Sent: Saturday, April 15, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail use external smtp server for sendingmail Yes, just configure your sendmail to do it. On 4/13/06, nik600 [EMAIL PROTECTED] wrote: is it possibile to set up an external smtp server for the relay to the users of the mails? 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 ___ --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
RE: [Asterisk-Users] multiple asterisk process ?
Yes it is a problem cos after a while of just leaving it the system is unable to make calls out via the PSTN, which is why I have spent time with the telco, more like wasted time, and played with zaptel's make options. After trying a few things I came to the temporary conclusion that it was the zaptel watchdog trying and failing to restart a hung channel. I recompiled zaptel without the watchdog and a few days later it did the same so I'm back to sq 1. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 18 April 2006 09:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] multiple asterisk process ? On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote: Any thoughts as to why only 1 of my boxes has this problem? Is it really a problem? I'm on a 2.6 kernel so any more ideas? Can someone answer what was the original purpose of the export LD_ASSUME_KERNEL=2.4.1 in the asterisk script? Perhaps Gregory Boehnlein, the author, will be able to tell us. -- Dave Cotton [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] multiple asterisk process ?
On Tue, 2006-04-18 at 09:33 +0100, Lee Archer wrote: Yes it is a problem cos after a while of just leaving it the system is unable to make calls out via the PSTN, which is why I have spent time with the telco, more like wasted time, and played with zaptel's make options. After trying a few things I came to the temporary conclusion that it was the zaptel watchdog trying and failing to restart a hung channel. I recompiled zaptel without the watchdog and a few days later it did the same so I'm back to sq 1. Ok, I'll ask it another way. Is it _the_ problem because I've an uptime of 209 days on a system with no problems and multiple asterisk processes. -- Dave Cotton [EMAIL PROTECTED] ___ --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 asterisk process ?
I've tried cat /proc/*asterisk proc number*/environ | strings | grep LD_ASSUME_KERNEL and it returns nothing..:( And just for confirmation : I had the same problem as Lee had (unable to make calls out) :( Regards, Stevanus Lee Archer wrote: Yes it is a problem cos after a while of just leaving it the system is unable to make calls out via the PSTN, which is why I have spent time with the telco, more like wasted time, and played with zaptel's make options. After trying a few things I came to the temporary conclusion that it was the zaptel watchdog trying and failing to restart a hung channel. I recompiled zaptel without the watchdog and a few days later it did the same so I'm back to sq 1. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave Cotton Sent: 18 April 2006 09:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] multiple asterisk process ? On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote: Any thoughts as to why only 1 of my boxes has this problem? Is it really a problem? I'm on a 2.6 kernel so any more ideas? Can someone answer what was the original purpose of the "export LD_ASSUME_KERNEL=2.4.1" in the asterisk script? Perhaps Gregory Boehnlein, the author, will be able to tell us. -- Dave Cotton [EMAIL PROTECTED] ___ --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 ___ --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] HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again
Hi all,I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1Everytime i press the music on hold button it seems that it stops music on hold and starts imediately again. Any one can guess what may be wrong?Best regards,Marco Mouta ___ --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: HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again
I forgot to write: When i hangup the call, it hangs correctly!On 4/18/06, Marco Mouta [EMAIL PROTECTED] wrote:Hi all,I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Stopped music on hold on Zap/1-1Everytime i press the music on hold button it seems that it stops music on hold and starts imediately again. Any one can guess what may be wrong?Best regards,Marco Mouta ___ --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 asterisk process ?
All I can figure is that something I haven't yet figured is causing these processes to be created, and after a while there is so many that outgoing calls over zap can't be made. It only applies to 1 system out of 7, running Suse 10 and a 2.6 kernel. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 18 April 2006 10:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] multiple asterisk process ? On Tue, 2006-04-18 at 09:33 +0100, Lee Archer wrote: Yes it is a problem cos after a while of just leaving it the system is unable to make calls out via the PSTN, which is why I have spent time with the telco, more like wasted time, and played with zaptel's make options. After trying a few things I came to the temporary conclusion that it was the zaptel watchdog trying and failing to restart a hung channel. I recompiled zaptel without the watchdog and a few days later it did the same so I'm back to sq 1. Ok, I'll ask it another way. Is it _the_ problem because I've an uptime of 209 days on a system with no problems and multiple asterisk processes. -- Dave Cotton [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] Sip.conf
Hi Check this setting: bindaddr = 0.0.0.0 :IP Address to bind to (listen on) kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Par?ina Sent: Tuesday, April 18, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sip.conf In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? -- Tomislav Parcina tparcina#lama.hr ___ --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] Grandstream Budgetone and Mac mini?
Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip.conf
In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? Try this in sip.conf under a phone definition: deny=0.0.0.0/0.0.0.0 permit=some_ip_address/some_mask Ivan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip.conf
2006/4/18, Tomislav Parčina [EMAIL PROTECTED]: In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? Asterisk can bind only ips from internal but I think the best way is to configure some firewall rules in your linux box. It is convenient to drop or reject all communications except that you want to accept (http, smtp, etc.). -- Alejandro Vargas ___ --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: Sip.conf
Hi Ivan! Thank you. I have already find it on http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask Problem is that they didn't define this in sip.conf.saple where I first have take a look. This should be fixed. -- Tomislav Parcina tparcina#lama.hr In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In sip.conf, how can I define that only IP phones from 192.168.0.0/24 network can register with specific user? The thing is that I can't use password and I can't use host=ip.of.my.phone. And I have to be sure that no one, from Internet will register on my * like that user. So, please tell me how to do this? Try this in sip.conf under a phone definition: deny=0.0.0.0/0.0.0.0 permit=some_ip_address/some_mask Ivan ___ --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: Sip.conf
Hi Alejandro! I have solved my problem. Look at mail above. One more thing, what you have suggested it's not an option. I have allowed that people from Internet can call me. So, I can't define any rule that will protect me the way you have mention. Thank you for trying. -- Tomislav Parcina tparcina#lama.hr Asterisk can bind only ips from internal but I think the best way is to configure some firewall rules in your linux box. It is convenient to drop or reject all communications except that you want to accept (http, smtp, etc.). -- Alejandro Vargas ___ --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: Quick question
Hi Alberto! Have you try it, do you use it? I ask because I was in contact with developers of ooh323 channel driver and they have told me that they can't grantee that it will work... I'm using oh323, version 0.67, and INBAND signalization doesn't work for me. I have an issue with version 0.73 that Asterisk won't load channel driver when it's starting... If you have more information's, please let me know. -- Tomislav Parcina tparcina#lama.hr In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You could try chan_oh323.so and chan_h323.so. I think also ooh323 supports inband DTMFs. Regards Alberto Sagredo ___ --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: Quick question
Hi Jeremy! I have noticed than almost nobody uses native H.323 driver. All that I have read about them is complaining that they don't support various stuff. Do you use h323 in production? -- Tomislav Parcina tparcina#lama.hr In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Tomislav Parčina wrote: Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! The native H.323 driver, chan_h323, does support inband DTMF. Good Luck, Jeremy McNamara ___ --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] Cisco 7970 SIP - few questions
- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is line button=4 featureID9/featureID ... For speeddial is line button=5 featureID2/featureID featureLabel341/featureLabel speedDialNumber341/speedDialNumber /line How to define hinting? - How to login true ssh? I have setup username and password, and when I try to log in it sends me challenge!?! login as: root [EMAIL PROTECTED]'s password: login: root challenge: YDXWGXMTpassword: Invalid Username/Password Entry. login: That is all, for now :)) -- Tomislav Parcina tparcina#lama.hr ___ --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 asterisk process ?
On 18 Apr 2006, at 09:27, Dave Cotton wrote: On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote: Any thoughts as to why only 1 of my boxes has this problem? Is it really a problem? I'm on a 2.6 kernel so any more ideas? Can someone answer what was the original purpose of the export LD_ASSUME_KERNEL=2.4.1 in the asterisk script? We have it set on Fedora Core 1 systems (and equivalent vintage RHEL) because RedHat backported the 2.6 kernel threads to their 2.4 kernel, and loads of things broke (oracle 9i, java 1.3 etc) so we had LD_ASSUME_KERNEL 2.4.1 set to force the old (i.e. normal for a 2.4 kernel) behavior. If you are running asterisk on a stable 2.6 kernel you shouldn't set it. By the way, the cat /proc/*asterisk proc number*/environ | strings | grep LD_ASSUME_KERNEL only works if you are root, or whoever asterisk is running as. It gives an empty result if it has not got permission to read. Tim Panton [EMAIL PROTECTED] ___ --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] Using ISDN MSNs for dialing out of Asterisk
Hi,
I am using Asterisk with misdn connected to an ISDN Line, so I have
several numbers I can use...
I know that I can use misdn like this in my extensions.conf:
exten = _0.,1,Dial(mISDN/1/${EXTEN:1})
But how can I use another number/MSN of my ISDN connection... it always uses
the default number, but i'd like to use another MSN for calling...
Can somebody help me please?
Thanks, Chris
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Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud
I'm running 1.2.3 and that seems to be the most stable version, had problems with other versions too. -- Original message -- From: "Brian Roy" [EMAIL PROTECTED] On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote: I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. I'm running 1.2.1 and most of mine get cut short too. I posted this on the list a few months ago and nobody had any suggestions. BJ said I should probably post a bug on it but I haven't had time to continue to troubleshoot it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been watching change logs and hadn't seen anything surrounding mixmonitor so I've let it go. Please continue to update us if anyone gets some resolution. I'm glad to know there are lots of us experiencing this. That should be the catalyst to get it fixed. -Brian ---BeginMessage--- ___ --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 ---End Message--- ___ --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] Granstream GXP2000 Distinctive tones
I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didnt seem to do the trick?? Has anyone one had any luck on this topic? Also havent been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the SPA 941's but no luck with the Grandstream units, I'm sure it can be done but not sure on the exact syntax to use Thanks in advance :) Cory ___ --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/FreePBX/Alcatel2400
Hello Fellow Users, This is my first post so please be kind :-) I am using asterisk@home with freePBX talking to an Alcatel 2400 analogue portvia a Digium TDM01B - 1 FXO card I can make calls into theasterisk by dialling theasterisk extn number from the 4200. Ican dial extns on the 4200 from asteriskby dialling 9(4200extn number) if I don't have any Outbound Dial prefix of 9 in my Trunk ZAP/g0 If I try to dial 9PSTN (90114xxx)number with this setting I get the autoattendant on my 4200 (the autoattendant is on Extn 0) If i put an Outbound Dial prefix of 9 in my Trunk ZAP/g0 I can dial 9PSTN (90114xx) but can no longer dial 4200 Extns Please could someone explain what i need to do to be able to call both Alcatel 4200 extns and external PSTN numbers from asterisk phones I only have one trunk/one outgoing routeset up at present Regards Andy GreenIT ManagerGBeyeLtd1 Russell StKelham IslandSheffieldS3 8RW Tel: 0114 252 1611Fax: 0114 272 9599 mailto:[EMAIL PROTECTED]http://www.businessgbeye.com Please read: CHANGE OF COMPANY NAME. As of 1st January 2006 GB Posters Ltd will be known as GB eye Ltd, please update all records and email addresses: Please replace everything after the @ in email addresses with gbeye.com (e.g. [EMAIL PROTECTED] is now [EMAIL PROTECTED]) The GB eye Ltd business website can be found at http://www.businessgbeye.com, please update your bookmarks and favourites. This e-mail is intended for the addressee(s) named above and any other use is prohibited. It may contain confidential information. If you received this e-mail in error please contact the sender by return e-mail. GB eye Ltd does not accept legal responsibility for the contents of this message if it has reached you via the Internet. Any opinions expressed are those of the author and are not necessarily endorsed by GB eye Ltd. Recipients are advised to apply their own virus checks to this message and all incoming email on delivery. ___ --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 calls dropout on PRI over TE110p
Try 1.2.3, it works fine. -- Original message -- From: "James Sturges" [EMAIL PROTECTED] I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once a day for 1 second, calls getting stuck, quite unpleasant! I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri. James-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul C Sent: Wednesday, 1 March 2006 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110pPaul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can makeoutgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ).SNIP overlapdial should usually be no in my experience.Okay I've turned that to no with no change. I've just got off the phone to Optus and apparently they had a client in melbourne last week and they fixed the problem by turning crc checking off at the optus end. I don't suppose that was anybody on here ? ___ & gt; -- 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] MixMonitor Problems -- sssshh, don't be too loud
See http://bugs.digium.com/view.php?id=6457 On Tue, 2006-04-18 at 13:11, [EMAIL PROTECTED] wrote: I'm running 1.2.3 and that seems to be the most stable version, had problems with other versions too. -- Original message -- From: Brian Roy [EMAIL PROTECTED] On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote: I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. I'm running 1.2.1 and most of mine get cut short too. I posted this on the list a few months ago and nobody had any suggestions. BJ said I should probably post a bug on it but I haven't had time to continue to troubleshoot it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been watching change logs and hadn't seen anything surrounding mixmonitor so I've let it go. Please continue to update us if anyone gets some resolution. I'm glad to know there are lots of us experiencing this. That should be the catalyst to get it fixed. -Brian __ From: Brian Roy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] MixMonitor Problems -- hh, don't be too loud Date: Sat, 04 Mar 2006 13:27:53 + ___ --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] Change email/pager VM alerts body text dynamically?
Greetings; I've got a situation where I really need to be able to select from a number of various (body) textx for email/pager messages when voivemails are left, but can't seem to figure out how to change them on the fly. Now, I know there are pre-defined defaults, which can be overridden by entries in the [general] section of voicemail.conf, but that does not get me there. I need to be able to select different message texts based upon the source of the voicemails. I tried creating additional voicemail 'contexts' to invoke, and added the emailsubject, pagersubject, etc. definitions there but that did not work. It would not seem to me that I should have to go to external code to accomplish this, so I expect that I'm just overlooking something obvious. (and will feel appropriately foolish whem someone points it out to me) But I'm stuck! Anyone out there made this happen? Thanks! -jim ___ --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] Grandstream Budgetone and Mac mini?
Just for shits and giggles, have you tried using a cross over cable? I'm not saying it's gonna work because everything I read says you're doing the right thing but it's worth a try. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dmitry Ivanov wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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] Granstream GXP2000 Distinctive tones
There is no way to do it currently on these phones. Although the web interface supports a distinctive ring based on callerID it does not accept wildcards. When I contacted Grandstream about this I got the following reply :- Unfortuantely, not with present firmware. We will implement this together with dial plan support. It is already in engineering agenda. Please keep tuned. Thank you very much for using our products. On Tue, 2006-04-18 at 13:14, Cory Hawkless wrote: I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn’t seem to do the trick?? Has anyone one had any luck on this topic? Also haven’t been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the SPA 941's but no luck with the Grandstream units, I'm sure it can be done but not sure on the exact syntax to use Thanks in advance :) Cory __ ___ --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] rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1Released)
Do your (wonderful) RPMs install also on CentOS? I suppose so because it is a Red Hat clone... Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Axel Thimm rpms for Fedora Core 1-5, RHEL 3-4 and RHL 7.3-9 have been updated: ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
Olivier Krief wrote: 2006/4/17, Nicholas Kathmann [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: I agree with Lee. I have about 30 machines in production using iaxmodem and hylafax which work perfectly. Most are running off of T1s, but some are on TDM400 and TDM2400s. I only use IBM servers (which are about twice the cost for the low end Dells), and have never had to resolve an IRQ problem. I just looked up the hylafax usage reports on those people running the analog FXOs, and one of them had 390 pages in the last week, only one error, which I would consider acceptable. Thanks, Nick 1. Do you mean Hylafax and Asterisk are installed on the machine and share the same TDM cards ? 2. If positive, do you have any extension which is used for both voice and fax ? For instance, user Alice receives voice or fax calls on its own extension. When it's a fax, your server detects it and and let Hylafax get the call. Cheers Olivier Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM). It uses iaxmodem to communicate between the two systems (imagine a softmodem). I'll create separate extensions for the iaxmodems, then either map the numbers (or channels off the TDM cards) to dial those extensions. You can also use the fax extension on your default incoming to dial the iaxmodem. Faxgetty then listens to the iaxmodem to receive faxes, and uses hylafax to send them to the appropriate email addresses, printers, etc. In most cases I'll set up separate PSTN numbers for incoming faxes, but the fax extension also works relatively well. The only time I've ever seen problems with faxes (or modems) is when trying to use a SIP or IAX provider over the internet. To connect the analog fax machines I'll either use a linksys PAP2 or Sipura SPA-2100. I used to use Grandstreams for that, but now find that they just randomly unregister themselves and have to be restarted before reconnecting. To do the scenario with both on the same extension, it would just be as follows: [incoming] s,1,However you handle your calls fax,1,Dial(IAX2/your iaxmodem extension) 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
[Asterisk-Users] Re: rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1Released)
On Tue, Apr 18, 2006 at 03:12:27PM +0200, Mimmus wrote: Do your (wonderful) RPMs install also on CentOS? I suppose so because it is a Red Hat clone... Yes, the RHEL builds will work on CentOS, WhiteBox, SL etc. by definition :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Axel Thimm rpms for Fedora Core 1-5, RHEL 3-4 and RHL 7.3-9 have been updated: -- Axel.Thimm at ATrpms.net pgplM22E5ofb7.pgp Description: PGP signature ___ --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] Grandstream Budgetone and Mac mini?
The PC port on a BT-102 should work with any computer that has an Ethernet card. Have you tried these phones with other computers than the Mac Minis you mention? It shouldn't make any difference whether the computer is a Mac, PC or anything else. Perhaps something is wrong with the BT-102s you have. -Rusty On 4/18/06, Dmitry Ivanov [EMAIL PROTECTED] wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Nicholas Kathmann wrote: Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM). It uses iaxmodem to communicate between the two systems (imagine a softmodem). I'll create separate extensions for the iaxmodems, then either map the numbers (or channels off the TDM cards) to dial those extensions. You can also use the fax extension on your default incoming to dial the iaxmodem. Faxgetty then listens to the iaxmodem to receive faxes, and uses hylafax to send them to the appropriate email addresses, printers, etc. In most cases I'll set up separate PSTN numbers for incoming faxes, but the fax extension also works relatively well. The only time I've ever seen problems with faxes (or modems) is when trying to use a SIP or IAX provider over the internet. To connect the analog fax machines I'll either use a linksys PAP2 I've found the same success here as well. TE110P, Asterisk. iaxmodem and HylaFAX. 332 pages in the last month with 1 failure. Had to tweak the rxgains a little, but afterwards, it just works. Thanks Lee and Steve! Doug ___ --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] voicemail kicking in after user has already disconnected
When someone hangs up before getting to the leave voicemail prompt, asterisk still attempts to record a voicemail message, so I end up getting a bunch of empty voicemails.. Is there any way to change this behaviour, so asterisk realizes that the channel has been disconnected, and does not attempt to record a voicemail message if this is the case? Thanks, Mike ___ --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] Grandstream Budgetone and Mac mini?
Are you seeing link on either end? Not sure if the GS shows or not. On the Mac, open a terminal window and type ifconfig to see if the port is active - ie has link it should have a line similar to this if so media: autoselect (100baseTX full-duplex) status: active If this is correct then you have something else wrong On Apr 18, 2006, at 8:22 AM, Rusty Dekema wrote: The PC port on a BT-102 should work with any computer that has an Ethernet card. Have you tried these phones with other computers than the Mac Minis you mention? It shouldn't make any difference whether the computer is a Mac, PC or anything else. Perhaps something is wrong with the BT-102s you have. -Rusty On 4/18/06, Dmitry Ivanov [EMAIL PROTECTED] wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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] Noise on IAX or SIP trunk between 2 Asterisk
Hi everyone: My escenario is: Meridian PBX (Connected to the PSTN) is connected to Asterisk-1 via PRI T1, the Asterisk 1 is connected to Asterisk 2 via IAX trunk or SIP trunk, the Asterisk 2 is used for predictive dialer with Answer Machine Detector, so for some reason, the AMD is starting at the moment the first ring back is detected, wich is not a normal behavior, because the AMD detect words. So finally i found the source of the problem: When the call is start to ring back generate some noise afecting to AMD. Any idea why the trunk or my configuration generate noise when the call is connected.? Regards. Cristian. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] voicemail kicking in after user has already disconnected
increase your silence setting On Apr 18, 2006, at 8:31 AM, Mike Garey wrote: When someone hangs up before getting to the leave voicemail prompt, asterisk still attempts to record a voicemail message, so I end up getting a bunch of empty voicemails.. Is there any way to change this behaviour, so asterisk realizes that the channel has been disconnected, and does not attempt to record a voicemail message if this is the case? Thanks, Mike ___ --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: Cisco 7940/7960 SIP 8.2 Freely
It doesn't seem as much broken as just annoying. I am holding off on upgrading until this resolves, but it doesn't seem to affect performance, anyways. BTW, some folks say that the server address only gets appended to the CID when a redirect or something comes about. Our experience here shows that the IP always gets appended. Alexander Burke wrote: Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: 8.2 isn't broken? Any comments? http://lists.digium.com/pipermail/asterisk-users/2006-March/143501.html Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phones that work well through NAT
So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, April 16, 2006 1:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones that work well through NAT I'm really not interested to look back, but IIRC, when using just one Polycom phone behind NAT we didn't have any problems, but when using more than one behind the same NAT that is when problems started, qualify=somethingbutno seemed to help it a bit, but didn't eliminate the problem. On 4/16/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 15 April 2006 22:37, C F wrote: That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Can you detail some problems? Just about any off-the-shelf router seems to work with these. There may be some cheap-ass broken routers you can get for $5 which will not work, but all of the brand-name stuff I've tried Just Works, which is why I say they work exceptionally well. -A. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/313 - Release Date: 4/15/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)
Andrew Kohlsmith wrote: On Monday 17 April 2006 07:44, Rich Adamson wrote: I don't believe you will ever get POTS - FXO-TDM400P-to-anything to work properly due to TDM card limitations. So, move all of those to the bottom of your list. I *had* this working. POTS - TDM400 TDM400 - Real_honest_fax_machine As I'd posted several times already. I have not been able to repeat this success, though. Same boat here. Certainly wish there was something that we could do to make it work as obviously there is a large market for the soho businesses that also need fax capability. R. ___ --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] Grandstream Budgetone and Mac mini?
I don't have any experience with that specific phone, but i have a little experience with Grandstream ATAs. Check web configuration to see is phone in gateway or bridge mode. It's just a guess since you haven't provide detailed info ;) What do you mean when you say that none of them works with mac mini? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: Tuesday, April 18, 2006 11:35 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream Budgetone and Mac mini? Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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 __ NOD32 1.1493 (20060417) Information __ This message was checked by NOD32 antivirus system. http://www.eset.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] IVR: playing multiple streams simultaneously?
Title: IVR: playing multiple streams simultaneously? Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help, Silviu ___ --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] Phones that work well through NAT
On Tuesday 18 April 2006 09:57, Sean Garland wrote: So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. You don't do anything to get it to work through NAT. If your * box is behind NAT you need to screw around a little, but for situations like this: * box --- [internet] --- [nat dsl router] --- IP501 all you do is set 'nat=yes' on the * box, in the IP501's peer setting. That's it. It even works with multiple IP501s behind the same NAT DSL router. If you have a stupid NAT box that closes ports off too quickly or plays too many games with the packets you may need some additional configuration (shorter registration expirations, etc.) but just buy a decent NAT box... WRT54Gs work just fine in their default configuration, for example. -A. ___ --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] Polycom Microbrowser
Hello, I read the polycom microbrowser post here http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser Can we access a webmail application like horde/imp or others (which ones) to read and listen voicemails , send e-mails, ... ? Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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] rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1Released)
On Tue, 2006-04-18 at 15:12 +0200, Mimmus wrote: Do your (wonderful) RPMs install also on CentOS? I suppose so because it is a Red Hat clone... There are asterisk 1.2.7.1 RPMs and SRPMs for CentOS 4.3 at http://www.laimbock/asterisk/ And as Axel already mentioned, there are asterisk RPMs and SRPMs for various platforms at his site at http://atrpms.net Regards, Patrick ___ --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] CAPI Installation Eicon Diva Server
Hi Avi This is great - the problem was how I configured my trunk so this part of your v. good wiki page was my solution: - Maximum channels: num of ports * 2 I have 2 ISDN lines active, so I have 4 maximum channels. If you have all 4 ports running, you have 8 maximum channels. Each ISDN line has 2 channels. Custom dial string: CAPI/g1/$OUTNUM$/b Alternatively, you could configure a trunk per port by using: CAPI/Contr1/$OUTNUM$/b You need to set 2 maximum channels for each port. - Too bad the documentation is a little sketchy on this stuff... Cheers, Nick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: 12 April 2006 22:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CAPI Installation Eicon Diva Server [EMAIL PROTECTED] wrote: Asterisk says it has 30 capi channels available, but my mistake may be in configuring the trunks... When I was debugging my Eicon Diva 4-BRI board, I found it useful to play with extensions_custom.conf (in AMP) just to ensure I got the Custom Dial String absolutely correct. According to the latest chan_capi-cm, the Dial String should be: CAPI/id/number/options Where: id = Contr1 or g1 (Controller or Group ID) number = Phone number options = Things like B or b for Early B3 and other things. I have 'b' in my options, but I do admit that I have no idea what early B3 is. :) Hope that helps in some way, Avi P.S. I wrote a quick config page for the 4-BRI for freePBX here: http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva It might have a few things to consider as well. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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 ** Any information in this communication which is confidential must not be disclosed to others without our consent. Such consent is not required where the information is publicly available and intended for onward distribution. If the information is confidential and if you are not the intended recipient, you are not authorised to and must not disclose, copy, distribute, or retain this message or any part of it. You are requested to return this message to the sender immediately. Due to the electronic nature of e-mail, there is a risk that the information contained in this message has been modified. Consequently Man Investments can accept no responsibility or liability as to the completeness or accuracy of the information. Visit us at: www.maninvestments.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] bad voice quality
Hi all, i have been having problems with voice quality. We run asterisk Asterisk CVS-HEAD version 2.4 as a production servre as a call centre/customer support engine. Pressenty, we have about 25 soft phones and 10 hard phones (Perfect Tone SIP phones). when handling internal calls, i usually notice a lot of static, echo and brakes (this is withing our local network) Our * runs on a supermicro P4SCi (3GHz) with 2GB memory running Debian Linuxkernel version2.6.12.3. Currently, we run gsm codec but we recently installed G729 codec (free version ). Please advice me on what to do and if more info is needed, i will be happy to provide more Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Doug Lytle wrote: Nicholas Kathmann wrote: Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM). It uses iaxmodem to communicate between the two systems (imagine a softmodem). I'll create separate extensions for the iaxmodems, then either map the numbers (or channels off the TDM cards) to dial those extensions. You can also use the fax extension on your default incoming to dial the iaxmodem. Faxgetty then listens to the iaxmodem to receive faxes, and uses hylafax to send them to the appropriate email addresses, printers, etc. In most cases I'll set up separate PSTN numbers for incoming faxes, but the fax extension also works relatively well. The only time I've ever seen problems with faxes (or modems) is when trying to use a SIP or IAX provider over the internet. To connect the analog fax machines I'll either use a linksys PAP2 I've found the same success here as well. TE110P, Asterisk. iaxmodem and HylaFAX. 332 pages in the last month with 1 failure. Had to tweak the rxgains a little, but afterwards, it just works. Thanks Lee and Steve! If you need to tweak gains something is seriously wrong. Steve ___ --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] IVR and voicemail issues ?
Hi, I have this setup in my extensions.conf: [inbound-analog] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(tag-welcome) exten = 1,1,Voicemail(u100) exten = 1,2,Hangup 'Zap/1-1' this means - press 1 and it goes to voicemail for extension 100. It all works well - except when you hangup. It does not clear the Zap/1-1 interface. It keeps it open - indefinitly. I read to use DeadAGI ? But could not figure this out - I am new to asterisk ;) Can anyone help ? Thanks Tonino ___ --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 code help
santosh y wrote: I'm very new to Asterisk, I'm tracing the Asterisk code, i'm feeling difficulty in understanding the code, so please tell me where i can get the documentation of the code and, design and architecture of the code. www.asteriskdocs.org is your best bet. Also sign up to svn-commits mailing list and hang around on irc. There is also a new developers conference which you may like to listen to. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Snom 360 Firmware 6.0 - Microbrowser
Hello, snom 360s can handle xml messages via SIP-Notify. Descriptions how to implement this on: http://snom.com/minibrowser/doc/xmlapplsnom360.pdf http://snom.com/minibrowser/notify.txt Common infos you can find out on: http://snom.com/wiki/index.php/Xmlobjects Hope this will help... cheers, Hirosh TWV wrote: By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware The XML minibrowser is very cool and opens a lot of possibilities! One of my ideas is rich messaging, so you can send fully formatted messages to a Snom 360 user! But... how can you make the phone navigate to a certain URL? (Initiated from the Asterisk side of course!) Is there some sort of SIP message or Asterisk Application / Command that can be used to make the phone browse to an xml URL? If not, this is a call to the nice people of Snom or the Asterisk community to add this functionality, it will be much needed! Thanks, Frederic ___ --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 -- snom technology AG Hirosh Dabui PGP Key-ID: 0x30A34758 mailto:[EMAIL PROTECTED] ___ --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] bad voice quality
There is a free version of G.729 available? I would be very interested in that!Alex On Mar Abr 18 15:50 , 'Dumpolid Exeplish' sent:Hi all, i have been having problems with voice quality. We run asterisk Asterisk CVS-HEAD version 2.4 as a production servre as a call centre/customer support engine. Pressenty, we have about 25 soft phones and 10 hard phones (Perfect Tone SIP phones). when handling internal calls, i usually notice a lot of static, echo and brakes (this is withing our local network) Our * runs on a supermicro P4SCi (3GHz) with 2GB memory running Debian Linux kernel version 2.6.12.3. Currently, we run gsm codec but we recently installed G729 codec (free version ). Please advice me on what to do and if more info is needed, i will be happy to provide more Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Steve Underwood wrote: Doug Lytle wrote: Nicholas Kathmann wrote: If you need to tweak gains something is seriously wrong. The 2 fax machines that I was having problem with were failing to train at 9600bps, they would then try at 7200 and finally train at 4800. Around 15 pages into the fax they would fail with a, Failed to detect high speed-data carrier and disconnect. Increasing the rxgain to 3.0 and they now train at 9600bps and faxes complete. This PRI is connected to our Definity G3. Doug ___ --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] Grandstream Budgetone and Mac mini?
The switch in the Budgetone is 10Base-T. If the PC NIC cannot auto-detect or otherwise handle that, it will be a problem. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Jerry Jones [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 18, 2006 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream Budgetone and Mac mini? Are you seeing link on either end? Not sure if the GS shows or not. On the Mac, open a terminal window and type ifconfig to see if the port is active - ie has link it should have a line similar to this if so media: autoselect (100baseTX full-duplex) status: active If this is correct then you have something else wrong On Apr 18, 2006, at 8:22 AM, Rusty Dekema wrote: The PC port on a BT-102 should work with any computer that has an Ethernet card. Have you tried these phones with other computers than the Mac Minis you mention? It shouldn't make any difference whether the computer is a Mac, PC or anything else. Perhaps something is wrong with the BT-102s you have. -Rusty On 4/18/06, Dmitry Ivanov [EMAIL PROTECTED] wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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] correct version of asterisk for oh323
Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Doug Lytle wrote: Steve Underwood wrote: Doug Lytle wrote: Nicholas Kathmann wrote: If you need to tweak gains something is seriously wrong. The 2 fax machines that I was having problem with were failing to train at 9600bps, they would then try at 7200 and finally train at 4800. Around 15 pages into the fax they would fail with a, Failed to detect high speed-data carrier and disconnect. Increasing the rxgain to 3.0 and they now train at 9600bps and faxes complete. This is probably the reason why in iaxmodem-0.1.3 this was done: make V.29 rx more sensitive (spandsp) 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] Asterisk Performance 350 Concurrent Channels Working Nicely
Hi All, This is a performance update. I have built appliance type servers with the following specs: Motherboard Asus P5MT-M Memory 1Gig DDR2 No hard drive, running in Ramdrive but using Sandisk Compact Flash to hold compressed image and /var directory Processor 3.2 Gig Pentium 4, HT Turned Off 2 on-board Gig NICs I'm using asterisk with looping call test configs to play audio and using 3 of the same spec servers to pound calls through 1 server. I managed to get 350 concurrent calls through with perfect audio consistently with ~20% idle processor load. Anything above that and things start breaking up. Using the latest 1.2.6 stable asterisk, I'm running into a limit of 276 SIP calls and no more. IAX calls can go 400+, so I test with combination 200+ SIP calls and the rest IAX and a combination of more and less SIP and IAX calls. Memory usage never goes over 256Meg, not sure why. Interesting, findings are very consistent with other performance testing that has been done over the years, Astertest and the like. HT turned on, SMB loaded in the kernel gave ~20% performance increase, BUT, using 425 + channels gave very inconsistent results, choppy audio, calls dropped, no audio, call setup time slowed. Good results below that mark, but not enough to warrant using full time. I'd rather build for stability and reliability than for all-out performance. Not too shabby, I'm very happy with this setup. 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
RE: [Asterisk-Users] correct version of asterisk for oh323
Hello, I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch versions of OpenH323 and Pwlib (available on http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php). Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] correct version of asterisk for oh323 Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
2006/4/18, Doug Lytle [EMAIL PROTECTED]: Nicholas Kathmann wrote: Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM).It uses iaxmodem to communicate between the two systems (imagine a softmodem).I'll create separate extensions for the iaxmodems, then either map the numbers (or channels off the TDM cards) to dial those extensions.You can also use the fax extension on your default incoming to dial the iaxmodem.Faxgetty then listens to the iaxmodem to receive faxes, and uses hylafax to send them to the appropriate email addresses, printers, etc.In most cases I'll set up separate PSTN numbers for incoming faxes, but the fax extension also works relatively well.The only time I've ever seen problems with faxes (or modems) is when trying to use a SIP or IAX provider over the internet.To connect the analog fax machines I'll either use a linksys PAP2Please, forgive my ignorance but could you elaborate how your system would be working ? I've read a lot about fax and Asterisk but I'm not sure I exactly got it (specially with iaxmodem and hylafax integration).Do you mean that :1. incoming calls would be routed according callee's extension (extensions are dedicated either to fax or voice applications) and only with that rule ? 2. you would exclusively connect existing fax machines to SIP ATA's and hope offering users the ability to fax from software applications, would decrease SIP ATA's use inconvenients ?CheersOlivier ___ --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] Voicemail problem
Hi, when I call the voicemail app, it starts and die suddenly. Has anyone already had this problem? Log: app.c:644 ast_play_and_record: No audio available on SIP/-6fca?? -- User hung up Tks, D.K. ___ --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] Help Getting Local Exchange Dialtone on PRI
Hi there,
i have a Problem with dialtone and a TE401P Card. I swear I surfed
the wiki, the mailing list and google for 4 hours and did not find the
solution, can you help me ?
In Germany I have an E1-Line and an Alcatel 4200 PRO PBX.
Without using asterisk I dial the 0 on an Alcatel Phone and have
the local exchange dialtone, then I can dial. Most users do not dial
en block, they dial number by number.
Now I have put an asterisk-server between the PRI and the Alcatel
and everything works fine and transparent, the only thing that is
missing is the dialtone after dialing 0.
My users are confused about this.
Do you have an idea how i can simulate the former behavior and provide the
local exchange dialtone to the user ? I have found out that i hear
the dialtone after a long time when i dial an empty number. But I
cannot use disa or something like that because the alcatel
phones don't do Tone-Dialing but some kind of inband dialing.
And I want en block dialing to also work because lots of applications dial
en block.
Basicaly I configured:
zapata.conf
[channels]
immediate=no
switchtype=euroisdn
overlapdial=yes
signalling=pri_cpe
..
group=1
context=telekom
signalling=pri_cpe
channel = 1-15,17-31
group=2
signalling=pri_net
context=alcatel
channel = 32-46,48-62
and in extensions.conf:
[telekom]
exten= _9149.,1,NoOp(Call from ${CALLERID} to ${EXTEN})
exten= _9149.,2,Dial(Zap/g2/${EXTEN:4})
exten= _9149.,3,Hangup
exten= _9149.,103,Playtones(busy)
exten= _9149.,104,Busy
[alcatel]
exten= _X.,1,NoOp(Call from ${CALLERID} to ${EXTEN})
exten= _X.,2,Dial(Zap/g1/${EXTEN})
exten= _X.,3,Hangup
exten= _X.,103,Playtones(busy)
exten= _X.,104,Busy
Thanks
Christoph
___
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Re: [Asterisk-Users] correct version of asterisk for oh323
Hi Herci, I have tried this. pwlib, openh323 and Asterisk-OH323 0.7.3 compiled with no problems. But when you start asterisk, Apr 18 17:47:39 ERROR[11385]: chan_oh323.c:5353 load_module: H.323 listener creation failed. Apr 18 17:47:39 WARNING[11385]: loader.c:414 __load_resource: chan_oh323.so: load_module failed, returning -1 == Cleaning up OpenH323 channel driver. Apr 18 17:47:39 WARNING[11385]: loader.c:554 load_modules: Loading module chan_oh323.so failed! I am using FC3 with 2.6.5-1.358 kernel. Any suggestions? yusuf Herchi Silviu wrote: Hello, I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch versions of OpenH323 and Pwlib (available on http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php). Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] correct version of asterisk for oh323 Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. ___ --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] Performance: Xeon or Opteron?
I have used many sangoma cards, and have not had *any* irq issues Anton Krall wrote: Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple cards? Problems with irq and such (same as with digium ones)? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Novack |Sent: Wednesday, April 12, 2006 10:29 AM |To: [EMAIL PROTECTED] |Cc: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron? | | | |Rich Adamson wrote: | | | While talking with one of the sangoma folks very recently, he was | rather emphatic the pci bus was designed to share |interrupts. I was | a little concerned as a test server had the wanpipe driver |sharing an | interrupt with libata and uhc1_hcd. His comment was that's the way | its suppose to work, sharing interrupts as needed. I've not had any | recognizable issues with the A200D card at all, and faxing |via a A200D | fxs port to a A200D fxo (pstn) port functions 100% reliably. | | What that would suggest is the TDM400 pci firmware (whether on card | logic or whatever) is the source of at least part of the |TDM400 shared | interrupt issue. I don't have any digium T1/E1 cards laying around, | but if memory serves correctly, the T1/E1 cards do not use the same | pci controller chip. That would suggest the T1/E1 cards are |less of an | issue then with the TDM400 card. | |That's good to know, but considering the response from Digium |on the TDM400 ( try another motherboard) when there didn't |seem to even be an int. sharing issue, the card just couldn't |be seen at all , and the support I received from Sangoma on a |recent FXS issue that was resolved within a few days, I would |tend to go with Sangoma for the T1 card, if and when I have the need. | |John Novack | |___ |--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 -- thanks, yusuf ___ --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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Olivier Krief wrote: 2006/4/18, Doug Lytle [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Nicholas Kathmann wrote: Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM). It uses iaxmodem to communicate between the two systems (imagine a softmodem). I'll create separate extensions for the iaxmodems, then either map the numbers (or channels off the TDM cards) to dial those extensions. You can also use the fax extension on your default incoming to dial the iaxmodem. Faxgetty then listens to the iaxmodem to receive faxes, and uses hylafax to send them to the appropriate email addresses, printers, etc. In most cases I'll set up separate PSTN numbers for incoming faxes, but the fax extension also works relatively well. The only time I've ever seen problems with faxes (or modems) is when trying to use a SIP or IAX provider over the internet. To connect the analog fax machines I'll either use a linksys PAP2 Please, forgive my ignorance but could you elaborate how your system would be working ? I've read a lot about fax and Asterisk but I'm not sure I exactly got it (specially with iaxmodem and hylafax integration). Do you mean that : 1. incoming calls would be routed according callee's extension (extensions are dedicated either to fax or voice applications) and only with that rule ? 2. you would exclusively connect existing fax machines to SIP ATA's and hope offering users the ability to fax from software applications, would decrease SIP ATA's use inconvenients ? Cheers Olivier The incoming call flow (for faxes) would be as follows: PSTN = TDM Card = Asterisk = iaxmodem = Hylafax = Email user (or printer) similarly, sending outgoing faxes can be as follows: User = Analog Fax = SIP ATA = Asterisk = TDM Card = PSTN or User = Hylafax (through email2fax or virtual fax printer) = iaxmodem = asterisk = TDM Card = PSTN (read the Hylafax docs to find out about sending faxes from hylafax) All of the incoming faxes don't actually go through the analog fax machine, rather through Hylafax directly to the user's email address or to a printer. Sending outgoing faxes can be done in either of the ways above. Once put in and the users are trained, the utilization of the analog fax machines generally goes down significantly. I've seen companies consolidate from 40+ analog faxes to less than 20 (large geographical area) with hylafax and asterisk. On your LAN you should be able to control voice quality issues to get reliable outgoing faxes. Incoming calls can either be routed through a separate DID for faxing, or through the same DID with what I showed you before. Asterisk is able to tell if the incoming call is a fax or voice, and send it to the fax extension (iaxmodem in this case) using the fax extension. You can run more than one instance of iaxmodem on a single machine, routing faxes to different email addresses from within hylafax depending on which extension they are received on. 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: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
Olivier Krief wrote: 2006/4/18, Doug Lytle [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Nicholas Kathmann wrote: Please, forgive my ignorance but could you elaborate how your system would be working ? I've read a lot about fax and Asterisk but I'm not sure I exactly got it (specially with iaxmodem and hylafax integration). Do you mean that : 1. incoming calls would be routed according callee's extension (extensions are dedicated either to fax or voice applications) and only with that rule ? I'm in a test environment at the moment, we've got a few clients that do large number of faxes. I'm testing with one of them. The machine is a Celeron 2.4ghz with 512MB memory, running Mandriva Linux. I've installed HylaFAX from source along with iaxmodem and Asterisk. iaxmodem is a software modem that uses the IAX protocol and registers to Asterisk as an IAX client allowing HylaFAX all the resources of the Asterisk PRI or whatever allows connectivity. HylaFAX's faxgetty program monitors the 23 iaxmodems that I run for incoming calls. On an inbound call (Being sent via our Definity G3) all caller information including DID is seen by HylaFAX and I can route to either printer/pdf or anything else via the FaxDispatch script. On an outbound call, HylaFAX sees the 23 iaxmodems as normal modems. Doug 2. you would exclusively connect existing fax machines to SIP ATA's and hope offering users the ability to fax from software applications, would decrease SIP ATA's use inconvenients ? We are currently trying to reduce the number of fax machines that we have. Hoping to centralize these functions in a multi-function server. We don't deal with ATAs. Doug ___ --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] BSR 1000 and Asterisk
HiI'm a new user of Asterisk and made the first VoIP call on my own LAN with a good quality. Now I want to configure a CMTS (motorola BSR 1000) and a server to support QoS. Does anyone knows how to configure this in order to work with SIP and with Asterisk? any ideas or tutorials? 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] Gizmo Call In
Hi, Anyone have a Call In number offered by Gizmo ( http://www.gizmoproject.com/call-in.php ) and have it configured in Asterisk sending and reciving calls to that number?. I have set one peer to Gizmo via SIP number provided by them but when I call to my Call In number assigned, calls arent routed to my Asterisk. Waiting any comment. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone and Mac mini?
Mac Ethernet ports are auto-switching. Don't need a cross-cable :) On 4/18/06, Mark Phillips [EMAIL PROTECTED] wrote: Just for shits and giggles, have you tried using a cross over cable? I'mnot saying it's gonna work because everything I read says you're doing the right thing but it's worth a try.Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.comDmitry Ivanov wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message
Hi, I'm trying to find how to configure Asterisk 1.2.7.1 to allow two EyeBeam (3015c) to send Instant Messages between them... But I cannot find anything that explains how to do it! Anybody as a clue? is it possible? Now, when we try to send an Instant Message in the eyeBeam it says: User not available. In asterisk console appears a message saying: -- Apr 18 17:13:22 WARNING[3473]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from JPAsip:[EMAIL PROTECTED];tag=4f2fd25b, dropped it... Content-Type:text/plain Message: this is a test --- (both users are online) Thanks, Joao Antunes ___ --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 Performance 350 Concurrent ChannelsWorking Nicely
Is this with Asterisk in the RTP stream? Is it doing any transcoding? -Original Message- From: JR Richardson [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 18, 2006 9:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent ChannelsWorking Nicely Hi All, This is a performance update. I have built appliance type servers with the following specs: Motherboard Asus P5MT-M Memory 1Gig DDR2 No hard drive, running in Ramdrive but using Sandisk Compact Flash to hold compressed image and /var directory Processor 3.2 Gig Pentium 4, HT Turned Off 2 on-board Gig NICs I'm using asterisk with looping call test configs to play audio and using 3 of the same spec servers to pound calls through 1 server. I managed to get 350 concurrent calls through with perfect audio consistently with ~20% idle processor load. Anything above that and things start breaking up. Using the latest 1.2.6 stable asterisk, I'm running into a limit of 276 SIP calls and no more. IAX calls can go 400+, so I test with combination 200+ SIP calls and the rest IAX and a combination of more and less SIP and IAX calls. Memory usage never goes over 256Meg, not sure why. Interesting, findings are very consistent with other performance testing that has been done over the years, Astertest and the like. HT turned on, SMB loaded in the kernel gave ~20% performance increase, BUT, using 425 + channels gave very inconsistent results, choppy audio, calls dropped, no audio, call setup time slowed. Good results below that mark, but not enough to warrant using full time. I'd rather build for stability and reliability than for all-out performance. Not too shabby, I'm very happy with this setup. 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 ___ --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] IVR: playing multiple streams simultaneously?
Title: IVR: playing multiple streams simultaneously? I have worked with several persons on this and there is currently an open request for sponsors for a whisper function. This is one of the features it will provide. Mixing streams to one channel. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herchi SilviuSent: Tuesday, April 18, 2006 10:29 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] IVR: playing multiple streams simultaneously? Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help, Silviu ___ --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] eyeBeam + ASterisk 1.2.7.1 + Instant Message
João Paulo Antunes wrote: Hi, I'm trying to find how to configure Asterisk 1.2.7.1 to allow two EyeBeam (3015c) to send Instant Messages between them... But I cannot find anything that explains how to do it! Anybody as a clue? is it possible? Now, when we try to send an Instant Message in the eyeBeam it says: User not available. In asterisk console appears a message saying: -- Apr 18 17:13:22 WARNING[3473]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from JPAsip:[EMAIL PROTECTED];tag=4f2fd25b, dropped it... Content-Type:text/plain Message: this is a test --- (both users are online) Thanks, Joao Antunes ___ --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 does not support sending messages like this at this moment. Sorry! -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --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] Voicemail Issue - Failed to lock path
What would cause this? It happened out of the blue: -- Executing VoiceMail(Zap/3-1, [EMAIL PROTECTED]) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') Apr 18 11:11:41 WARNING[15841]: app.c:1164 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/326/INBOX': File exists Apr 18 11:11:41 ERROR[15841]: app_voicemail.c:5569 vm_exec: Could not leave voicemail. The path is already locked. Apr 18 11:11:41 WARNING[15841]: app_voicemail.c:5573 vm_exec: Extension 3326, priority 103 doesn't exist. -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (internal, 3326, 3) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] correct version of asterisk for oh323
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Tuesday, April 18, 2006 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] correct version of asterisk for oh323 Hi Herci, I have tried this. pwlib, openh323 and Asterisk-OH323 0.7.3 compiled with no problems. But when you start asterisk, Apr 18 17:47:39 ERROR[11385]: chan_oh323.c:5353 load_module: H.323 listener creation failed. Apr 18 17:47:39 WARNING[11385]: loader.c:414 __load_resource: chan_oh323.so: load_module failed, returning -1 == Cleaning up OpenH323 channel driver. Apr 18 17:47:39 WARNING[11385]: loader.c:554 load_modules: Loading module chan_oh323.so failed! I am using FC3 with 2.6.5-1.358 kernel. Any suggestions? yusuf Herchi Silviu wrote: Hello, I've used Asterisk 1.2.6 and Asterisk-OH323 0.7.3 with the Mimas patch versions of OpenH323 and Pwlib (available on http://www.inaccessnetworks.com/projects/asterisk-oh323). It all works OK except for the CallerID bug in Asterisk-OH323 0.7.3 (see https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php). Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: mardi 18 avril 2006 17:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] correct version of asterisk for oh323 Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. ___ --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 Yusuf, I used the same OS; I am able to run it with no problem. I used the pwlib and openh323 libs from http://www.voxgratia.org/downloads.html Pwlib-Mimas_patch2-src-tar.gz Openh323-Mimas_patch2-src-tar.gz Asterisk-oh323-0.7.3.tar.gz I compiled then in the order and load it on asterisk 1.2.5 and its working fine. Goksie ___ --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 crash with Digium
Hi. I have a problem with two asterisk servers with version 1.2.5. In one server there is a Digium TE411P in the second the Digium TE100P. We use E1 and EuroISDN. '/etc/zaptel.conf': - begin - span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 - end - /etc/asterisk/zapata.conf': - begin - [channels] pridialplan = unknown prilocaldialplan = international localprefix = 0512 nationalprefix = 0 internationalprefix = 00 overlapdial = yes switchtype = euroisdn usecallerid = yes immediate = no echocancel = yes callerid = asreceived signalling = pri_cpe context = incoming group = 1 channel = 1-15,17-31 - end - On both systems we have irregularly asterisk crashes without a log file or something else. Update to a higher asterisk version is not possible, because we got regular function problems with an error like the change to line 2 does not work, because the line is busy etc. We have also many asterisk-servers in different versions active, all without a digium card, they all work fine. If it is useful to determine the error we can send a core dump file for analysis. Thank you very much. Best regards. -- Rudolf E. Steiner [EMAIL PROTECTED] ___ --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] Double Ring - TelIAX/Cisco 79[46]0
Anyone experience the double ringing when calling out over TelIAX? I am using a Cisco 79[46]0, and do not use the r option in the Dial() command. I always thought that the r is what causes double ring, and is never really needed except to cause problems... Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.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] T1 to cross connect remote PBX and asterisk
Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an asterisk box providing PRI_Network signaling on a T1 interface card using a long haul point to point ESF/B8ZS T1? I do not need the technical details on how to set up asterisk or the remote PBX, just need a sanity check on the idea of using the PTP T1 as a cross connect facility. If they were local to each other I would simply drop in a T1 crossover cable, but they are not J 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] ISDN in Japan?
Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with ...INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 so does that mean I can't use a Digium card it there? Can someone please clarify what sort of system I'm looking at here and if I need a japanese retailer for the card or what ;-) Thanks! -- Chris Earle System Solutions Specialist -- 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] Polycom Microbrowser
Because of the small screen real estate you might want to use something like squirrelmail from squirrelmail.org -- this would require some chopping up to make it fit anyway, but might be easier to implement then imp moj [EMAIL PROTECTED] wrote: Hello, I read the polycom microbrowser post here http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser Can we access a webmail application like horde/imp or others (which ones) to read and listen voicemails , send e-mails, ... ? Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] ISDN in Japan?
On Tue, 18 Apr 2006, Chris Earle (CBL) wrote: Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with ...INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 so does that mean I can't use a Digium card it there? Can someone please clarify what sort of system I'm looking at here and if I need a japanese retailer for the card or what I don't know the status of ISDN in Japan, but the Eicon DIVA Server cards (BRI and PRI) are provided with firmware for ISDN protocols in japan. Together with chan-capi it is fully functional with Asterisk/OpenPBX. Armin ___ --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] eyeBeam + ASterisk 1.2.7.1 + Instant Message
I don't think Asterisk supports SIP MESSAGE, does it? -Original Message- From: João Paulo Antunes [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 18, 2006 10:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message Hi, I'm trying to find how to configure Asterisk 1.2.7.1 to allow two EyeBeam (3015c) to send Instant Messages between them... But I cannot find anything that explains how to do it! Anybody as a clue? is it possible? Now, when we try to send an Instant Message in the eyeBeam it says: User not available. In asterisk console appears a message saying: -- Apr 18 17:13:22 WARNING[3473]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from JPAsip:[EMAIL PROTECTED];tag=4f2fd25b, dropped it... Content-Type:text/plain Message: this is a test --- (both users are online) Thanks, Joao Antunes ___ --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] Aastra 9133i Phones Asterisk 1.2.6 and MWI
Hi, I have several aastra 9133i phones, which are connected to an asterisk 1.2.6 system. I have setup MWI on the phones to point to the IP of the asterisk server, but although there is a message waiting new in the mailbox, the phone's light does not light. Any thoughts? ___ --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 Performance 350 Concurrent
Asterisk was in the RTP and no transcoding, straight Ulaw g.711. -- 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] PRI blocking on incoming calls
Ok here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using RedFones FoneBridges. We also have a Nortel Option 11C that we have hooked up to the Asterisk. We have 3 T1s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four are configured with em_w. We are, at this point, only using Asterisk as an IVR with plans to move off the Nortel in the future if we can make this work. We have a second FoneBridge with four PRIs connected to our Nortel 11C using esf and b8zs and pri_net. The telco T1s do not have D-Channels but the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are played a message about call recording and then the call is transferred to the Nortel system to be processed by an agent. When we first fire this up all seems to work just fine, calls come in, get the message and then transfer to the Nortel and on to an agent. Everybody is happy. The problem is after 5-20 minutes calls on the MCI lines start getting busy signals. The Qwest line NEVER stops working. We would place a few test calls on the MCI and get busy signals and then they start going through again. A few minutes later they get busy signals again. When we get the busy signals there is no response on the Asterisk CLI with verbose at 10. Its as if the Asterisk is not ever seeing the call. What is annoying is that it works fine for a bit and then starts hiccupping. Can anyone shed any light on where to look? Any help would be desperately appreciated. Please help. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] T1 to cross connect remote PBX and asterisk
In theory it should work, I have just written a proposal for someone that involves such a setup. in worst case I can always utilize the circuit as data, and use 2 asterisk boxes one on each end to convert it back to NI2. On 4/18/06, Damon Estep [EMAIL PROTECTED] wrote: Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an asterisk box providing PRI_Network signaling on a T1 interface card using a long haul point to point ESF/B8ZS T1? I do not need the technical details on how to set up asterisk or the remote PBX, just need a sanity check on the idea of using the PTP T1 as a cross connect facility. If they were local to each other I would simply drop in a T1 crossover cable, but they are not J 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 ___ --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] T1 to cross connect remote PBX and asterisk
Yep, I agree, Just watch out for regulatory issues if you are in the USA, handing a CUSTOMER a TDM interface vs. a SIP/VoIP interface falls under a much different regulatory and jurisdictional set of rules... Have you talked to anyone that has confirmed an implementation like described works without issues? Damon -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, April 18, 2006 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 to cross connect remote PBX and asterisk In theory it should work, I have just written a proposal for someone that involves such a setup. in worst case I can always utilize the circuit as data, and use 2 asterisk boxes one on each end to convert it back to NI2. On 4/18/06, Damon Estep [EMAIL PROTECTED] wrote: Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an asterisk box providing PRI_Network signaling on a T1 interface card using a long haul point to point ESF/B8ZS T1? I do not need the technical details on how to set up asterisk or the remote PBX, just need a sanity check on the idea of using the PTP T1 as a cross connect facility. If they were local to each other I would simply drop in a T1 crossover cable, but they are not J 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 ___ --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] Dialplan - strip IDD prefix and insert another
How's tou're service with Sellvoip, I was not able to intergrate them into my system and they had no phone support. I'm using Gafachi now but prefer the rates Sellvoip provide. -- Original message -- From: "AR Tarzi" [EMAIL PROTECTED] SellVoIP appears to follow a US dialplan. A US numberis dialled as 1NXXNXX whereas an international (to the US) numberis dialled as 011X. Frankly, I didn't ask whether international numbers like Barbados where the code remains as 1 butare international (to the US) need the 011 or can be dialled directly but that's not really my concern. I've assumed they don't. Most of the world uses 00 as the internation prefix code, therefore I have to ask: Howcan I "strip" the 00 and insert 011 in one entry in the dialplan. I'm stripping the 00 and passing the rest of the numbers for numbersdialled as001X. (as in: 00|1XX.) but in case of numbers out of the US, how would I insert the 011 ? ---BeginMessage--- BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAy== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060305T155750Z END:VCARD
RE: [Asterisk-Users] PRI blocking on incoming calls
I believe it is important to determine if the issue arrives at TDMoE level (RedPhone uses it to avoid a direct T1 link with the box ) Or at PRI level. I am not clear what kink of link bridges your redPhone to Asterisk. Is it an ethernet link or a T1 crossover? Ing. Oscar Andrés Carriles Presidente InFoDaX Consultants Nicolás Jorge 994 (B1706AVA) Haedo Buenos Aires, Argentina Tel: 54 11 4650 1775 Fax: 54 11 4650 4295 www.infodax.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy Enviado el: Martes, 18 de Abril de 2006 03:30 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] PRI blocking on incoming calls Ok here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using RedFones FoneBridges. We also have a Nortel Option 11C that we have hooked up to the Asterisk. We have 3 T1s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four are configured with em_w. We are, at this point, only using Asterisk as an IVR with plans to move off the Nortel in the future if we can make this work. We have a second FoneBridge with four PRIs connected to our Nortel 11C using esf and b8zs and pri_net. The telco T1s do not have D-Channels but the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are played a message about call recording and then the call is transferred to the Nortel system to be processed by an agent. When we first fire this up all seems to work just fine, calls come in, get the message and then transfer to the Nortel and on to an agent. Everybody is happy. The problem is after 5-20 minutes calls on the MCI lines start getting busy signals. The Qwest line NEVER stops working. We would place a few test calls on the MCI and get busy signals and then they start going through again. A few minutes later they get busy signals again. When we get the busy signals there is no response on the Asterisk CLI with verbose at 10. Its as if the Asterisk is not ever seeing the call. What is annoying is that it works fine for a bit and then starts hiccupping. Can anyone shed any light on where to look? Any help would be desperately appreciated. Please help. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 ___ --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] Realtime goto problem
Hi, Sample database ++---+---+--+-+- -+ | id | context | exten | priority | app | appdata | ++---+---+--+-+- -+ | 1 | incoming| 6069 |1 | Goto| incoming-next|6069|1 | | 2 | incoming| 6069 |2 | Hangup | | | 3 | incoming-next | 6069 |1 | DigitTimeout| 10 | | 4 | incoming-next | 6069 |2 | ResponseTimeout | 30 | | 5 | incoming-next | 6069 |3 | Background | welcome If i dont declare the incoming-next context in extensions.conf I get: Channel 'Zap/21-1' sent into invalid extension '1' in context 'incoming-next ', but no invalid handler. But if I put on extensions.conf: [incoming-next] Switch = Realtime/@ ,it works fine. Do we need to declare all contexts in extensions.conf so we can use it on Realtime?? Another question: Its possible to include contexts in Realtime like we made on extensions.conf? Thanks in advance, Pedro Nunes ___ --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: Sixtel Services
there 2 types of inbound metered and unmetered. unmetered is unlimited inbound and metered charges per the minutes. -- Original message -- From: "Steve Totaro" [EMAIL PROTECTED] Inbound should be free as far as I am concerned unless you have a toll free number. Thanks, Steve Totaro _ From: VIC IP Communications [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 11:28 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] re: Sixtel Services Hi, Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly, do these companies provide inbound and outbound routing of calls, or are these rates strictly for inbound Call routing of DIDs? & gt; 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 ___ --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 redundancy
I will tell you straight up that NFS mounted volumes will cause asterisk to croak if it needs access to something that's not mounted. The first time the NFS share disappears for a moment, you're going to be restarting services and losing time on the asterisk machines that need the mounts. It would be better to drop the files on all the systems so you don't have to worry about that. Agreed, just having the servers mount the NFS for MWI makes asterisk fall to its knees. Was thinking of using distributed file system. Anyone ever give CODA a try (or any of the other distributed file systems) ? But definetly parts of the internal asterisk DB needs to be shared between the servers. Ben ___ --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] T1 to cross connect remote PBX and asterisk
I have our Avaya connected to Asterisk using NI D channel protocol over a standard ESF/B8ZS span. It works great. Pretty easy. On Asterisk's side I just had to tell it: in zapata.conf: [channels]switchtype=nationalsignalling=pri_cpegroup=1channel = 1-23 in zaptel.conf: loadzone= usdefaultzone= usspan=1,0,0,esf,b8zsbchan=1-23dchan=24 Jim From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: Tuesday, April 18, 2006 11:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] T1 to cross connect remote PBX and asterisk Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an asterisk box providing PRI_Network signaling on a T1 interface card using a long haul point to point ESF/B8ZS T1? I do not need the technical details on how to set up asterisk or the remote PBX, just need a sanity check on the idea of using the PTP T1 as a cross connect facility. If they were local to each other I would simply drop in a T1 crossover cable, but they are not J Thanks! This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] PRI blocking on incoming calls
We have a crossover from telco to the CSU and a crossover from the CSU to the RedFone and then a regular Ethernet cable from the RedFone to the Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles Sent: Tuesday, April 18, 2006 2:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PRI blocking on incoming calls I believe it is important to determine if the issue arrives at TDMoE level (RedPhone uses it to avoid a direct T1 link with the box ) Or at PRI level. I am not clear what kink of link bridges your redPhone to Asterisk. Is it an ethernet link or a T1 crossover? Ing. Oscar Andrés Carriles Presidente InFoDaX Consultants Nicolás Jorge 994 (B1706AVA) Haedo Buenos Aires, Argentina Tel: 54 11 4650 1775 Fax: 54 11 4650 4295 www.infodax.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy Enviado el: Martes, 18 de Abril de 2006 03:30 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] PRI blocking on incoming calls Ok here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using RedFones FoneBridges. We also have a Nortel Option 11C that we have hooked up to the Asterisk. We have 3 T1s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four are configured with em_w. We are, at this point, only using Asterisk as an IVR with plans to move off the Nortel in the future if we can make this work. We have a second FoneBridge with four PRIs connected to our Nortel 11C using esf and b8zs and pri_net. The telco T1s do not have D-Channels but the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are played a message about call recording and then the call is transferred to the Nortel system to be processed by an agent. When we first fire this up all seems to work just fine, calls come in, get the message and then transfer to the Nortel and on to an agent. Everybody is happy. The problem is after 5-20 minutes calls on the MCI lines start getting busy signals. The Qwest line NEVER stops working. We would place a few test calls on the MCI and get busy signals and then they start going through again. A few minutes later they get busy signals again. When we get the busy signals there is no response on the Asterisk CLI with verbose at 10. Its as if the Asterisk is not ever seeing the call. What is annoying is that it works fine for a bit and then starts hiccupping. Can anyone shed any light on where to look? Any help would be desperately appreciated. Please help. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 ___ --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
