[Asterisk-Users] limiting incloming call on sip phones to 1
Hey Guys! I know sip hpones can be configured to disable call waiting but this is for all call appearances. I was wondering if there is a way to limit outgoing calls (asterisk - phone) to a sip phone (techonology) to 1? Is there any other way of doing this without groups or such? Any kind of settings on sip.conf for this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
Brand new cards, from a recommended Digium distributor. What rev? It'll say Freshmaker Rev x on the blue board. I *think* it was corrected around Rev E or F IIRC, but I don't know for certain. I spoke with Digium and the distributor tech support on it - gave them serial numbers, board rev numbers, etc. No one mentioned this. Probably because your boards are Rev I or later, which I believe are the current boards. FWIW, Rev J is current as of Oct 2005, but it is functionaly equivalent to Rev I. Difference involves some line filtering required by certain country stanadards and use of rj11's instead of rj45 jacks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to conferencd in Asterisk
Hi all How ro enable conference in asterisk and also how to make call between sccp device and sip device is there any special config in asterisk. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 403 Forbidden
Hi - I'm new to Asterisk and I've got asterisk@ home running with a TDM12B with Grandstream 2000 phones. I can login to the phones using the browser, login to the AMP, but I can't make any calls either internal or external, via softphone or Grandstream. I get an error 403 Forbidden in the soft phone. The Grandstream gives a busy signal and 503 error or 403 on line3. The calls are being logged in the AMP as NO ANSWER. I just sent this in to Digium, but I don't think it's a related issue. Thanks for any ideas - Tim eMail to Digium: I'm getting an error that indicates 1 of my 3 cards on the TDM12B is not working. I bought it from Digium Canada. -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Unable to do INITIAL ProSLIC powerup on module 0 ProSLIC on module 0 failed to powerup within 510 ms (0 mV only) -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Unable to do INITIAL ProSLIC powerup on module 0 Module 0: FAILED FXS (FCC) Module 1: Not installed Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) Registered tone zone 0 (United States / North America) usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) ip_tables: (C) 2000-2002 Netfilter core team sis900.c: v1.08.06 9/24/2002 divert: allocating divert_blk for eth0 eth0: Realtek RTL8201 PHY transceiver found at address 1. eth0: Using transceiver found at address 1 as default eth0: SiS 900 PCI Fast Ethernet at 0x8800, IRQ 9, 00:0c:6e:0d:1e:10. ip_tables: (C) 2000-2002 Netfilter core team eth0: Media Link On 100mbps full-duplex parport0: PC-style at 0x378 (0x778) [PCSPP,TRISTATE] parport0: irq 7 detected lp0: using parport0 (polling). lp0: console ready usb.c: registered new driver serial usbserial.c: USB Serial support registered for Generic usbserial.c: USB Serial Driver core v1.4 audit subsystem ver 0.1 initialized lspci 00:00.0 Host bridge: Silicon Integrated Systems [SiS] 651 Host (rev 02) 00:01.0 PCI bridge: Silicon Integrated Systems [SiS] Virtual PCI-to-PCI bridge (AGP) 00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS962 [MuTIOL Media IO] (rev 25) 00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] Sound Controller (rev a0) 00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) 00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) 00:03.3 USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller 00:04.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI Fast Ethernet (rev 91) 00:0e.0 Network controller: Unknown device e159:0001 01:00.0 VGA compatible controller: ATI Technologies Inc Radeon RV250 If [Radeon 9000] (rev 01) 01:00.1 Display controller: ATI Technologies Inc Radeon RV250 [Radeon 9000] (Secondary) (rev 01) [ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Conference-reg
Hi all I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 extension.conf [default] exten = 1234,1,Meetme(1234) pls do the needful.. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 403 Forbidden
Hi - I'm new to Asterisk and I've got asterisk@ home running with a TDM12B with Grandstream 2000 phones. I can login to the phones using the browser, login to the AMP, but I can't make any calls either internal or external, via softphone or Grandstream. I get an error 403 Forbidden in the soft phone. The Grandstream gives a busy signal and 503 error or 403 on line3. The calls are being logged in the AMP as NO ANSWER. I just sent this in to Digium, but I don't think it's a related issue. Thanks for any ideas - Tim eMail to Digium: I'm getting an error that indicates 1 of my 3 cards on the TDM12B is not working. I bought it from Digium Canada. -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Have you answered the question above? That's telling you that you didn't plug in a power connector to the edge of the TDM card when you installed it. Your Granstream phones are likely not working because apparently you have not configured them to register with asterisk. You will somehow need to program the phones with a userid and password so then can register with asterisk, and, you will need to tell asterisk what the userid and passwords are for those phones (so it can authorize the registration). Since this is an Asterisk user's list (and not an Asterisk at Home list), you'd probably have better luck getting appropriate responses from the asterisk at home list. The majority of folks on this list don't use asterisk at home so can't offer a lot of suggestions. I'd have to guess there are instructions/guidelines or something with asterisk at home that you've probably not read as yet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Conference-reg
I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 I assume you put the above in meetme.conf file? extension.conf [default] exten = 1234,1,Meetme(1234) Is the [default] section of extensions.conf where all of your other extensions are defined? If not, move the above entry to whatever section you have your other extensions defined. Then stop and restart asterisk. If the above doesn't address your issue, then copy/paste the CLI stuff so we can see what it is telling you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
Andrew Kohlsmith wrote: On Saturday 05 November 2005 22:33, Gary Eck wrote: I have popping with FSO modules only on channel 1 - the other 3 channels are clear. That was corrected a long time ago. You must have an older rev TDM400 carrier card. -A. Well, it must be back. I had two rev I TDM400 cards in two different servers and port one on both boards would crack, pop, etc. I called Digium and was told they had a bad batch of cards. They RMA'd mine and two new ones are on the way. I wish I had called Digium before I spent two days pursuing interrupt problems :) Cheers, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
Rich Adamson wrote: Brand new cards, from a recommended Digium distributor. What rev? It'll say Freshmaker Rev x on the blue board. I *think* it was corrected around Rev E or F IIRC, but I don't know for certain. I spoke with Digium and the distributor tech support on it - gave them serial numbers, board rev numbers, etc. No one mentioned this. Probably because your boards are Rev I or later, which I believe are the current boards. FWIW, Rev J is current as of Oct 2005, but it is functionaly equivalent to Rev I. Difference involves some line filtering required by certain country stanadards and use of rj11's instead of rj45 jacks. That's wierd. I have two rev I TDM400's and both have rj11 jacks. As a matter of fact, that caused me a bit of a problem. I was used to rj45. I made up a bunch of patch cables for an install an used cat5 w/rj45. Tried to plug them in and guess what ... did't fit :) Cheers, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limiting incloming call on sip phones to 1
Anton Krall wrote: Hey Guys! I know sip hpones can be configured to disable call waiting but this is for all call appearances. I was wondering if there is a way to limit outgoing calls (asterisk - phone) to a sip phone (techonology) to 1? Is there any other way of doing this without groups or such? Any kind of settings on sip.conf for this? ___ You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / CVS head) in sip.conf for that extension. These limits are named from asterisk's perspective. incominglimit is calls coming in to asterisk, so it would limit calls from the sip phone to asterisk, but not from asterisk to the phone. outgoinglimit (1.0.x) doesn't work from what I've read. call-limit is both directions. It may be what you need. However, you won't be able to do an attended transfer. Blind transfer might work, but I haven't tried it. quote from previous thread from Olle Johansson: incominglimit is replaced by call-limit. Please read sip.conf.sample. Outgoinglimit has not worked for ages, so we removed it. One limit works for both incoming and outgoing calls now. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Conference-reg
Have you checked your zaptel interface. If you don't have hardware then use ztdummy. I guess you would have. ~Vamsi On 11/6/05, nr k [EMAIL PROTECTED] wrote: Hi allI am having Asterisk 1.0.9. now i configured themeetme conference with conference number 1234 and alsoi add the extension 1234 in extension.conf.if i callto 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices.[room]; Usage is conf = confno[,pin]conf = 1234extension.conf[default]exten = 1234,1,Meetme(1234)pls do the needful..regards ramakrishnan.n__Yahoo! Mail - PC Magazine Editors' Choice 2005http://mail.yahoo.com___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uninstall AMP
Wow that was mean. -J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian C. Fertig Sent: Friday, November 04, 2005 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Uninstall AMP rm rf / ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uninstall AMP Hi! How do I uninstall AMP and FOP from my Asterisk? Regards Anders Svensson This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] limiting incloming call on sip phones to 1
Hi Kebin., Thx for your comments, their exactly what I read. Problem comes when you want to be able to make any number of incoming calls (calls from the phone out) but limit the number of outgoing calls (calls from asterisk to the phone). :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin Hanson |Sent: Sunday, November 06, 2005 9:09 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] limiting incloming call on sip |phones to 1 | |Anton Krall wrote: | |Hey Guys! | |I know sip hpones can be configured to disable call waiting |but this is |for all call appearances. I was wondering if there is a way to limit |outgoing calls (asterisk - phone) to a sip phone (techonology) to 1? | |Is there any other way of doing this without groups or such? Any kind |of settings on sip.conf for this? | | |___ | | |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / |CVS head) in sip.conf for that extension. | |These limits are named from asterisk's perspective. |incominglimit is calls coming in to asterisk, so it would |limit calls from the sip phone to asterisk, but not from |asterisk to the phone. outgoinglimit (1.0.x) doesn't work |from what I've read. | |call-limit is both directions. It may be what you need. |However, you won't be able to do an attended transfer. Blind |transfer might work, but I haven't tried it. | |quote from previous thread from Olle Johansson: | |incominglimit is replaced by call-limit. Please read sip.conf.sample. | |Outgoinglimit has not worked for ages, so we removed it. One |limit works for both incoming and outgoing calls now. | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP error setting up AMP
The /etc/asterisk directory has the same permissions. Any other ideas? Thanks again, Dave - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 06, 2005 12:00 AM Subject: Re: [Asterisk-Users] PHP error setting up AMP On Sat, Nov 05, 2005 at 02:18:00PM -0700, David D. Dixon wrote: I've previously run AAH (installing from the ISO and tar), but this time I'm doing my own install and am having problems getting AMP to work right. Any time I try to modify the configuration, I get an error like this: Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission denied in /var/www/html/admin/functions.php on line 2292 There was a post to this list back in March that was asking about a similar problem, and it was said to be related to file permissions on the /etc/asterisk folder and the conf files in it. I have chmod-ed the folder and the files in it, so that all users have rwx permissions: -rwxrwxrwx 1 asterisk asterisk 1695 Oct 13 2004 agents.conf Any ideas on what my problem might be? I had to manually add the asterisk user that apache runs with, but did not create a password. If this could be part of the problem, where would I find out what the PHP scripts/Asterisk are using? What about the directory /etc/asterisk itself? BTW: generally you should solve permissions problems with ghown/chgrp and not with chmod 777. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNS Server Failure wreaks havoc
I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: * Is it true that if I hand-resolve the server names in all the config files, and then use those instead of the hostnames, this problem won't occur? That's not exactly optimal, of course, since it defeats the whole purpose of dynamic name binding. But it's hard to explain to my SOHO customers, who don't really need any IP-based functionality (although I give all of them some complimentary minutes on nufone) why their phones go down when the Internet is down. * Is it true that there's no way to get applications in Linux, generally speaking, to try more than a single server when doing a name resolve? Only the first server listed in /etc/resolv.conf (on my systems, anyway) seems to ever get consulted. I think both of these situations are pretty serious failings, if in fact they're failings in the systems and not this bedeviled cranium. Thanks. B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP support is making good progress
On Mon, Oct 31, 2005 at 09:27:27AM -, Chris Bagnall wrote: This is a old firmware issue, upgrading the phone firmware everything is working ok with the 7960 Sadly, that's the problem at the moment - I can't seem to get hold of new firmware for love nor money. Even the hunting for firmware on ebay route yielded zero results when I had a look yesterday. I know this is wrong but try looking on edonkey serach cmterm for sccp (cmterm-7940-7960-sccp.7-2-3.zip) or P0S3-07-5-00 for SIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re-invite don't always work
I want to be SURE that two UAs connected by asterisk (1.2-beta2) use a direct RTP stream, so that they don't waste the bandwidth of asterisk. How can I obtain it? I have set canreinvite=yes, but I have read that in this case asterisk TRY to do a reinvite, but if it don't succeed, it remains in the middle. Is it right? Looking at the output of a tcpdump it seems that actually it doesn't work in any condition. We have a Cisco PSTN gateway that calls the asterisk, witch forward the call to one of two phones. In the case of an analog phone attached to a Fritz! Box Fon WLAN, it seems that the RTP stream don't flow through asterisk. In the case of a Grandstream GXP-2000, it seams that it sends its RTP stream directly to the gateway BUT the gateway keeps sending its RTP stream through asterisk! Anybody knows why it happens? How can I avoid this? How can i FORCE asterisk to ALWAYS reinvite the calls? I prefer the call to NOT be established instead of flowing through asterisk. Thanks. -- ___ __ |- [EMAIL PROTECTED] |ederico Giannici http://www.neomedia.it ___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] escaping to an extension while listening to voicemail message
Guys. I was wondering, some voicemail systems let you escape to another extension or context while listening to the voicemail greeting, for example, for leaving faxes, like Hi, you have reached XXX, if you want to leave a fax, press 5 now, otherwise stay to leave voicemail. Can this be done on asterisk? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Asterisk is horrible at handleing DNS failures. Don't use DNS with Asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re-invite don't always work
Federico Giannici wrote: How can i FORCE asterisk to ALWAYS reinvite the calls? I prefer the call to NOT be established instead of flowing through asterisk. You can't. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] escaping to an extension while listening to voicemail message
Anton Krall wrote: Guys. I was wondering, some voicemail systems let you escape to another extension or context while listening to the voicemail greeting, for example, for leaving faxes, like Hi, you have reached XXX, if you want to leave a fax, press 5 now, otherwise stay to leave voicemail. Can this be done on asterisk? See show application voicemail Pay special attention to the notes about the o and a extensions. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limiting incloming call on sip phones to 1
You can try the SetGroup CheckGroup apps to do that. On 11/6/05, Anton Krall [EMAIL PROTECTED] wrote: Hi Kebin., Thx for your comments, their exactly what I read. Problem comes when you want to be able to make any number of incoming calls (calls from the phone out) but limit the number of outgoing calls (calls from asterisk to the phone). :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin Hanson |Sent: Sunday, November 06, 2005 9:09 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] limiting incloming call on sip |phones to 1 | |Anton Krall wrote: | |Hey Guys! | |I know sip hpones can be configured to disable call waiting |but this is |for all call appearances. I was wondering if there is a way to limit |outgoing calls (asterisk - phone) to a sip phone (techonology) to 1? | |Is there any other way of doing this without groups or such? Any kind |of settings on sip.conf for this? | | |___ | | |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / |CVS head) in sip.conf for that extension. | |These limits are named from asterisk's perspective. |incominglimit is calls coming in to asterisk, so it would |limit calls from the sip phone to asterisk, but not from |asterisk to the phone. outgoinglimit (1.0.x) doesn't work |from what I've read. | |call-limit is both directions. It may be what you need. |However, you won't be able to do an attended transfer. Blind |transfer might work, but I haven't tried it. | |quote from previous thread from Olle Johansson: | |incominglimit is replaced by call-limit. Please read sip.conf.sample. | |Outgoinglimit has not worked for ages, so we removed it. One |limit works for both incoming and outgoing calls now. | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] escaping to an extension while listening tovoicemail message
The operator and * extensions? Let me check that out. Ok, I guess I can tie something up to the * extension Thx for the tip Eric. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Eric ManxPower Wieling |Sent: Sunday, November 06, 2005 11:59 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] escaping to an extension while |listening tovoicemail message | |Anton Krall wrote: | Guys. | | I was wondering, some voicemail systems let you escape to another | extension or context while listening to the voicemail greeting, for | example, for leaving faxes, like Hi, you have reached XXX, if you | want to leave a fax, press 5 now, otherwise stay to leave voicemail. | | Can this be done on asterisk? | |See show application voicemail Pay special attention to the |notes about the o and a extensions. |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] limiting incloming call on sip phones to 1
Will give it a run.. Thx CF.. Ill check the wiki for examples. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of C F |Sent: Sunday, November 06, 2005 12:37 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] limiting incloming call on sip |phones to 1 | |You can try the SetGroup CheckGroup apps to do that. | |On 11/6/05, Anton Krall [EMAIL PROTECTED] wrote: | Hi Kebin., | | Thx for your comments, their exactly what I read. Problem comes when | you want to be able to make any number of incoming calls (calls from | the phone | out) but limit the number of outgoing calls (calls from asterisk to | the phone). | | :( | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of Kevin | |Hanson | |Sent: Sunday, November 06, 2005 9:09 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] limiting incloming call on sip phones | |to 1 | | | |Anton Krall wrote: | | | |Hey Guys! | | | |I know sip hpones can be configured to disable call waiting | |but this is | |for all call appearances. I was wondering if there is a |way to limit | |outgoing calls (asterisk - phone) to a sip phone |(techonology) to 1? | | | |Is there any other way of doing this without groups or such? Any | |kind of settings on sip.conf for this? | | | | | |___ | | | | | |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / |CVS head) | |in sip.conf for that extension. | | | |These limits are named from asterisk's perspective. | |incominglimit is calls coming in to asterisk, so it would |limit calls | |from the sip phone to asterisk, but not from asterisk to |the phone. | |outgoinglimit (1.0.x) doesn't work from what I've read. | | | |call-limit is both directions. It may be what you need. | |However, you won't be able to do an attended transfer. Blind | |transfer might work, but I haven't tried it. | | | |quote from previous thread from Olle Johansson: | | | |incominglimit is replaced by call-limit. Please read |sip.conf.sample. | | | |Outgoinglimit has not worked for ages, so we removed it. One limit | |works for both incoming and outgoing calls now. | | | |___ | |--Bandwidth and Colocation sponsored by Easynews.com -- | | | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | ___ | --Bandwidth and Colocation sponsored by Easynews.com -- | | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem ringing multiple extensions when one is forwarded
We have an incoming line which rings a large group of phones. If one of the phones is set to call-forward, the entire group is diverted. We would like asterisk to ignore the forward and continue to ring the rest of the phones. Any ideas how this could be done? I suspect that ring groups could be used to solve this problem but the documentation is very light in this area. By the way, the phones are Cisco 7912s 7940s and the forwarding is set on the phones themselves, not in asterisk. Regards, -- John Lange ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP extension calls itself intermittently
Thank you very much for the help! I continue to have the mystery calls but not as often. I have attached the debugging info I captured. I also removed every piece of equipment and have a single line coming from the NIU into the X100P Clone card. I do have DSL so a DSL filter is required along is in the middle. What actually causes Asterisk/ZAP to thick there is a call? Thanks, Chip Nov 6 08:31:12 VERBOSE[4851]: -- Starting simple switch on 'Zap/1-1' Nov 6 08:31:20 WARNING[4851]: CallerID returned with error on channel 'Zap/1-1' Nov 6 08:31:20 VERBOSE[4851]: -- Executing Dial(Zap/1-1, SIP/6000|20) in new stack Nov 6 08:31:20 DEBUG[4851]: Setting NAT on RTP to 0 Nov 6 08:31:20 DEBUG[4851]: Outgoing Call for 6000 Nov 6 08:31:20 DEBUG[4851]: Call from user '6000' is 1 out of 0 Nov 6 08:31:20 VERBOSE[4851]: -- Called 6000 Nov 6 08:31:21 VERBOSE[4851]: -- SIP/6000-0821 is ringing Nov 6 08:31:28 DEBUG[4851]: update_user_counter(6000) - decrement outUse counter Nov 6 08:31:28 DEBUG[4851]: Exiting with DIALSTATUS=CANCEL. Nov 6 08:31:28 VERBOSE[4851]: == Spawn extension (from-pstn, s, 1) exited non-zero on 'Zap/1-1' Nov 6 08:31:28 VERBOSE[4851]: -- Executing Hangup(Zap/1-1, ) in new stack Nov 6 08:31:28 VERBOSE[4851]: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1' Nov 6 08:31:28 DEBUG[4851]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Nov 6 08:31:28 DEBUG[4851]: disabled echo cancellation on channel 1 Nov 6 08:31:28 DEBUG[4851]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Nov 6 08:31:28 DEBUG[4851]: Updated conferencing on 1, with 0 conference users Nov 6 08:31:28 VERBOSE[4851]: -- Hungup 'Zap/1-1' Nov 6 08:31:28 DEBUG[4851]: Acked pending invite 102 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, November 05, 2005 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP extension calls itself intermittently Intermittently Ill get calls from my only SIP extension to itself via the Zap/1. I have no clue and have found nothing online. I have listed my configurations and a sample of the console messages I see why debugging. Right now it only happens to the 6000 extension. Any assistance is appreciated. [from-pstn] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,Dial(SIP/6000,20) exten = s,4,Voicemail(u6000) exten = s,5,Congestion exten = s,6,Hangup In the above, you don't want to answer and incoming call in your dialplan. When SIP/6000 picks up the phone, an answer is automatically sent back to the pstn. So, change the above to something like this: [from-pstn] exten = s,1,Dial(SIP/6000,20) exten = s,2,Voicemail(u6000) exten = s,102,Voicemail(b6000) exten = s,103,Hangup The following file is /etc/asterisk/zapata.conf, not zaptel.conf zaptel.conf [channels] language=en context=from-pstn switchtype=national busydetect=yes busycount=4 callprogress=yes signalling=fxs_ks rxwink=300 usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=name (xxx) xxx- busydetect=yes busycount=4 callprogress=yes channel = 1 Far too much junk and duplication in the above. Read the following including the comments. [channels] language=en busydetect=yes busycount=4 cidsignalling=bell cidstart=ring callerid=asreceived ; everything listed above applies to all channels defined below. Therefore ; only have to define them one time. ; the following channel definition is for the pstn line (channel 1) context=from-pstn ; switchtype=national ; this statement is for ISDN, not analog pstn. remove it ; busydetect=yes ; these two statements belong above and apply to all channels. ; busycount=4 callprogress=yes ; this should probably be =no signalling=fxs_ks ; rxwink=300 ; this statement isn't used with fxs_ks, remove it. usecallerid=yes ; cidsignalling=bell ; these two statements belong above and apply to all channels. ; cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ; not sure about the statement. remove it. callwaitingcallerid=yes threewaycalling=yes transfer=yes; the pstn caller is _not_ going to transfer anything. remove it. cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes ; these two statements are identical. remove one of them. echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 ; this statement isn't needed, remove it. pickupgroup=1 ; the pstn line isn't going to pick up a call. remove it. immediate=no callerid=name (xxx) xxx- ; not needed. remove it. callprogress=yes ; used this
Re: [Asterisk-Users] Problem ringing multiple extensions when one is forwarded
I'm not so sure that the entire group is diverted. Lets see first: 1. How are you calling these phones? 2. Are you using Zap? 3. If the forward is to a local extensions, does the same thing happen? Also please post your CLI output. For some reason I think you are using Zap channels, and the Cisco phone is forwarded to an external number that uses a Zap FXO port, which to asterisk is answered as soon as it starts dialing, the workaround might be to put a c in the dial coommand, which requires a confirmation when the phone rings to be considered answred. Or you could simply block phone enabled forwards that involve using Zap FXO ports. On 11/6/05, John Lange [EMAIL PROTECTED] wrote: We have an incoming line which rings a large group of phones. If one of the phones is set to call-forward, the entire group is diverted. We would like asterisk to ignore the forward and continue to ring the rest of the phones. Any ideas how this could be done? I suspect that ring groups could be used to solve this problem but the documentation is very light in this area. By the way, the phones are Cisco 7912s 7940s and the forwarding is set on the phones themselves, not in asterisk. Regards, -- John Lange ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] upgrade to 1.2 beta 2 issue
Ever since I upgraded to beta2, the console is littered with these kind of messages: NOTICE[206]: chan_iax2.c:5654 update_registry: Restricting registration for peer 'kkai13' to 60 seconds (requested 0) Any way to suppress this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem ringing multiple extensions when one is forwarded
Here is a bit more information. First, to clarify, you are correct the entire group is not diverted, however, since the forward is going to a direct to voicemail extension it answers immediately and that stops the group from ringing. What we need is for asterisk to completely ignore the forward and just ring the remaining phones. BTW, pri_gw below is a Cisco sip gateway connected to a PRI. Here is a sanitized mini-version of the CLI output. -- Executing Dial(SIP/10.0.0.1-b5437768,SIP/EXTEN1SIP/EXTEN2SIP/EXTEN3.. etc. -- Called EXTEN1 -- Called EXTEN2 -- Called EXTEN3 -- SIP/EXTEN1-848c is ringing -- SIP/EXTEN2-8d15 is ringing -- Got SIP response 302 Moved Temporarily back from 10.0.0.56 -- Now forwarding SIP/10.0.0.1-b5437768 to 'Local/forwardnum@context' (thanks to SIP/EXTEN3-7642) -- Executing SetCallerID(Local/forwardnum@context-81d5,2, Name 2021234567) in new stack -- Executing Dial(Local/forwardnum@context-81d5,2, SIP/forwardnum@pri_gw) in new stack -- Called forwardnum@pri_gw -- SIP/pri_gw-6f36 is making progress passing it to Local/forwardnum@context-81d5,2 -- Local/forwardnum@context-81d5,1 is making progress passing it to SIP/10.0.0.1-b5437768 -- SIP/pri_gw-6f36 answered Local/forwardnum@context-81d5,2 -- Local/forwardnum@context-81d5,1 answered SIP/10.0.0.1-b5437768 -- Attempting native bridge of SIP/10.0.0.1-b5437768 and SIP/pri_gw-6f36 Thanks, John On Sun, 2005-11-06 at 14:23 -0500, C F wrote: I'm not so sure that the entire group is diverted. Lets see first: 1. How are you calling these phones? 2. Are you using Zap? 3. If the forward is to a local extensions, does the same thing happen? Also please post your CLI output. For some reason I think you are using Zap channels, and the Cisco phone is forwarded to an external number that uses a Zap FXO port, which to asterisk is answered as soon as it starts dialing, the workaround might be to put a c in the dial coommand, which requires a confirmation when the phone rings to be considered answred. Or you could simply block phone enabled forwards that involve using Zap FXO ports. On 11/6/05, John Lange [EMAIL PROTECTED] wrote: We have an incoming line which rings a large group of phones. If one of the phones is set to call-forward, the entire group is diverted. We would like asterisk to ignore the forward and continue to ring the rest of the phones. Any ideas how this could be done? I suspect that ring groups could be used to solve this problem but the documentation is very light in this area. By the way, the phones are Cisco 7912s 7940s and the forwarding is set on the phones themselves, not in asterisk. Regards, -- John Lange ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream HandyTone 386 HT386 Distinctive Ring with _ALERT_INFO
Doesn't seem that these ATA devices support the Set(_ALERT_INFO=bellcore-dr[1-8]) directive, or Classic-$num that I've seen in a few posts. Is it possible for * to set a distinctive ring to these devices? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and cisco ubr900 configs using h.323.
I was wondering if anyone has the working configs for asterisk h323.conf and for the cisco ubr900 voip box? TIA, Todd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ABE - Are you happy with it?
Can any one who has gone from the open source version of Asterisk to ABE comment on their experiences? Specifically: - How does the quality compare to the open source stable versions? - How often do updates come out? - How far is it behind CVS HEAD in terms of features? - How good had Digium support been? - Overall was the switch worth the money? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
I agree about Asterisk being terrible with DNS failure, but how can you avoid using DNS on *nix system?On 11/7/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before.I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down.The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable.I have two questions; being able to solve either would be wonderful:Asterisk is horrible at handleing DNS failures.Don't use DNS with Asterisk.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Why don't you cache the DNS locally. That way if the remote DNS goes down you'll only lose updates. -- 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 sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] escaping to an extension while listening tovoicemail message
The 'o' works well - especially with the attended transfer function. PaulH - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 4:58 AM Subject: Re: [Asterisk-Users] escaping to an extension while listening tovoicemail message Anton Krall wrote: Guys. I was wondering, some voicemail systems let you escape to another extension or context while listening to the voicemail greeting, for example, for leaving faxes, like Hi, you have reached XXX, if you want to leave a fax, press 5 now, otherwise stay to leave voicemail. Can this be done on asterisk? See show application voicemail Pay special attention to the notes about the o and a extensions. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re-invite don't always work
And some phones have a faulty re-invite that doesn't work. (they are listed on the wiki) PaulH - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 4:57 AM Subject: Re: [Asterisk-Users] Re-invite don't always work Federico Giannici wrote: How can i FORCE asterisk to ALWAYS reinvite the calls? I prefer the call to NOT be established instead of flowing through asterisk. You can't. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re-invite don't always work
And some phones have a faulty re-invite that doesn't work. (they are listed on the wiki) PaulH - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 4:57 AM Subject: Re: [Asterisk-Users] Re-invite don't always work Federico Giannici wrote: How can i FORCE asterisk to ALWAYS reinvite the calls? I prefer the call to NOT be established instead of flowing through asterisk. You can't. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with SIP Phones inside a NAT with * inside another NAT
Hi List... I was hunting for a solution to a small issue I have been having and came across a Page on the Brilliant http://www.voip-info.org site... At the bottom of this page:- http://www.voip-info.org/wiki/view/SER+tips+and+tricks There is an item in the wishlist titled.. Complete example to setup SER [EMAIL PROTECTED] on a single machine (i.e. allow remote SIP extensions behind home gateway NAT devices to connect seamlessly). Has anyone been able to come up with an answer to this yet ? If anyone can help me or point me in the correct direction please let me know on or off-list... Thank You Best Regards Gavin Spurgeon Assistant Systems Administrator [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
- Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 8:11 AM Subject: Re: [Asterisk-Users] DNS Server Failure wreaks havoc Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Why don't you cache the DNS locally. That way if the remote DNS goes down you'll only lose updates. Maybe even run the DNS cache on the Asteirsk box itself. PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limiting incloming call on sip phones to 1
If the phone has two lines on it, you can be creative and set them up differently. (one for incoming, no limit. one for outgoing, limited to 1) PaulH - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 3:35 AM Subject: RE: [Asterisk-Users] limiting incloming call on sip phones to 1 Hi Kebin., Thx for your comments, their exactly what I read. Problem comes when you want to be able to make any number of incoming calls (calls from the phone out) but limit the number of outgoing calls (calls from asterisk to the phone). :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin Hanson |Sent: Sunday, November 06, 2005 9:09 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] limiting incloming call on sip |phones to 1 | |Anton Krall wrote: | |Hey Guys! | |I know sip hpones can be configured to disable call waiting |but this is |for all call appearances. I was wondering if there is a way to limit |outgoing calls (asterisk - phone) to a sip phone (techonology) to 1? | |Is there any other way of doing this without groups or such? Any kind |of settings on sip.conf for this? | | |___ | | |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / |CVS head) in sip.conf for that extension. | |These limits are named from asterisk's perspective. |incominglimit is calls coming in to asterisk, so it would |limit calls from the sip phone to asterisk, but not from |asterisk to the phone. outgoinglimit (1.0.x) doesn't work |from what I've read. | |call-limit is both directions. It may be what you need. |However, you won't be able to do an attended transfer. Blind |transfer might work, but I haven't tried it. | |quote from previous thread from Olle Johansson: | |incominglimit is replaced by call-limit. Please read sip.conf.sample. | |Outgoinglimit has not worked for ages, so we removed it. One |limit works for both incoming and outgoing calls now. | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP extension calls itself intermittently
Thank you very much for the help! I continue to have the mystery calls but not as often. I have attached the debugging info I captured. I also removed every piece of equipment and have a single line coming from the NIU into the X100P Clone card. I do have DSL so a DSL filter is required along is in the middle. What actually causes Asterisk/ZAP to thick there is a call? Thanks, Chip Nov 6 08:31:12 VERBOSE[4851]: -- Starting simple switch on 'Zap/1-1' Nov 6 08:31:20 WARNING[4851]: CallerID returned with error on channel 'Zap/1-1' Don't know. Have you had the line tested by the telco? In the mean time, try busycount=8 to see if that might be a short term solution while waiting on the telco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Aterisk 1.2.0 beta 2 and sip dtmf
I have a number of sipura 2002 ATA connected to asterisk. I have set them up with 'dtmfmode=rfc2833' for handling dtmf. I then setup setup an extension to test sending dtmf tones. exten = *40,1,Answer exten = *40,n,Wait(2) exten = *40,n,SendDTMF(123456789,500) exten = *40,n,Hangup When a Call the extension I get a bunch of very short clicks but not the valid dtmf tones. I then use ethereal to look at what is being set I got the following 3 packets RFC2833 DTMF event packet that where the same bu the sequence number increrment by 1. timestamp 0 Sequence number incrment by 1 for each end event 0 volume 10 duration0 followed by 3 RFC2833 DTMF event packet that were the same. Timestamp 0 Sequence number same for all 3 end event 1 volume 10 duration800 This does not look at all correct to me. Should it have sent TimeStamp 0 Sequence Number end Event 1 volume 10 duration 800 or TimeStamp 0 Sequence Number end Event 0 volume 10 durationx then Timestamp x sequece Number +1 end Event 1 volume 10 duration (number of count left) Now a couple of question about the 3 events Why does asterisk send 3 RFC2833 events with the same timestamp for 1 start of tone ? Why is the duration set to zero at the start (is the min value for duration 40 ms not zero) ? Why does asterisk send end event with the same timecode as the start event ? Is there a reason not to send 1 DTMF event with the end bit sent and the duration correct ? As a note this play ok on SIPURA 841 but not 3k or 2002. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stuck getting SIP trunk to work with 404 error.
Hi,
I have been trying all kinds of combinations, but still no luck getting my SIP
trunk from my asterisk to work. Asterisk says it is registered with
pim*CLI sip show registry
HostUsername Refresh State
sip.voicedata.be:5060 3199118004 100 Registered
And it is working when connected directly to my sip phone.
But when I try to dail some number on the internet, the phone returns a 404
error, it will however call all internal numbers.
Hope someone can help to fix tis problem.
Below the config.
sip.conf
[general]
context=default ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is
5060)
;bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
bindaddr=193.172.54.90 ; Listen only to the unique address.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;nat=yes; NAT settings
register = 3199118004:[EMAIL PROTECTED]
[sip_voicedata.be]
type=friend
context=directdial
secret=
username=319911
fromuser=319911
host=sip.voicedata.be
insecure=very
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
[from-sip]
exten = 1001,1,Dial(SIP/1001)
exten = 1001,2,Congestion
[directdial]
ignorepat = 9
exten = _906.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
exten = _906.,2,Congestion
[default]
include = from-sip
;include = directdial
___
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Re: [Asterisk-Users] DNS Server Failure wreaks havoc
Um, put in IP addresses instead of hostnames in Asterisk's config files? Eric Bishop wrote: I agree about Asterisk being terrible with DNS failure, but how can you avoid using DNS on *nix system? On 11/7/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Asterisk is horrible at handleing DNS failures. Don't use DNS with Asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
Eric ManxPower Wieling wrote: Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Asterisk is horrible at handleing DNS failures. Don't use DNS with Asterisk. I have found Asterisk is terrible with any kind of internet outage. IAX stops trying to register if the internet goes down for a few minutes and the customer looses long distance calling until a tech resets the PBX. I had to setup a cron job to reload asterisk every hour to get any kind of reliability. I also found the same problem with local calls, when I lose internet the locals calls go out also. How can that make any sense? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
DNS caching server running in the same machine ? Eric ManxPower Wieling wrote: Um, put in IP addresses instead of hostnames in Asterisk's config files? Eric Bishop wrote: I agree about Asterisk being terrible with DNS failure, but how can you avoid using DNS on *nix system? On 11/7/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Asterisk is horrible at handleing DNS failures. Don't use DNS with Asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
Chris Mason wrote: I also found the same problem with local calls, when I lose internet the locals calls go out also. How can that make any sense? Doesn't make any sense, but when I reported that to some geek-head friends a couple of months back, they didn't believe me. I don't like it, but I'm sorta glad to find there are others. I was wondering about my Mental Health. B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Testing with X101P
Hi there, im testing my asterisk box using a Modem Intel 56K which on the documentation says it must have the same behavior as an X101P. So im trying to configure just a simple line with 6 extensions. Asterisk loads fine and when im testing an incoming call the welcome message answers but when im trying to dial to any extension, anything happens is like asterisk dont recognice any digit after the welcome. Im using Asterisk version 1.0.9 and im attaching my extensions.conf which is very simple. Any clue will be very helpful. Thanks a lot for your help. Carlos Andres Medina __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com extensions.conf Description: 3949034846-extensions.conf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] escaping to an extension while listeningtovoicemail message
Can you post an example? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Sunday, November 06, 2005 3:32 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] escaping to an extension while |listeningtovoicemail message | |The 'o' works well - especially with the attended transfer function. | |PaulH | |- Original Message - |From: Eric ManxPower Wieling [EMAIL PROTECTED] |To: Asterisk Users Mailing List - Non-Commercial Discussion |asterisk-users@lists.digium.com |Sent: Monday, November 07, 2005 4:58 AM |Subject: Re: [Asterisk-Users] escaping to an extension while |listening tovoicemail message | | | Anton Krall wrote: | Guys. | | I was wondering, some voicemail systems let you escape to another |extension | or context while listening to the voicemail greeting, for |example, for | leaving faxes, like Hi, you have reached XXX, if you want |to leave a |fax, | press 5 now, otherwise stay to leave voicemail. | | Can this be done on asterisk? | | See show application voicemail Pay special attention to the notes | about the o and a extensions. | ___ | --Bandwidth and Colocation sponsored by Easynews.com -- | | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] limiting incloming call on sip phones to 1
This by using setgroups? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Sunday, November 06, 2005 5:03 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] limiting incloming call on sip |phones to 1 | |If the phone has two lines on it, you can be creative and set |them up differently. |(one for incoming, no limit. one for outgoing, limited to 1) | |PaulH | |- Original Message - |From: Anton Krall [EMAIL PROTECTED] |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |asterisk-users@lists.digium.com |Sent: Monday, November 07, 2005 3:35 AM |Subject: RE: [Asterisk-Users] limiting incloming call on sip |phones to 1 | | | Hi Kebin., | | Thx for your comments, their exactly what I read. Problem |comes when you | want to be able to make any number of incoming calls (calls |from the phone | out) but limit the number of outgoing calls (calls from |asterisk to the | phone). | | :( | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of | |Kevin Hanson | |Sent: Sunday, November 06, 2005 9:09 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] limiting incloming call on sip | |phones to 1 | | | |Anton Krall wrote: | | | |Hey Guys! | | | |I know sip hpones can be configured to disable call waiting | |but this is | |for all call appearances. I was wondering if there is a |way to limit | |outgoing calls (asterisk - phone) to a sip phone |(techonology) to 1? | | | |Is there any other way of doing this without groups or |such? Any kind | |of settings on sip.conf for this? | | | | | |___ | | | | | |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b / | |CVS head) in sip.conf for that extension. | | | |These limits are named from asterisk's perspective. | |incominglimit is calls coming in to asterisk, so it would | |limit calls from the sip phone to asterisk, but not from | |asterisk to the phone. outgoinglimit (1.0.x) doesn't work | |from what I've read. | | | |call-limit is both directions. It may be what you need. | |However, you won't be able to do an attended transfer. Blind | |transfer might work, but I haven't tried it. | | | |quote from previous thread from Olle Johansson: | | | |incominglimit is replaced by call-limit. Please read |sip.conf.sample. | | | |Outgoinglimit has not worked for ages, so we removed it. One | |limit works for both incoming and outgoing calls now. | | | |___ | |--Bandwidth and Colocation sponsored by Easynews.com -- | | | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | ___ | --Bandwidth and Colocation sponsored by Easynews.com -- | | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Testing with X101P
Hello Carlos, Try putting in a exten = s,5,WaitExten,5 after your background,welcome. It will give you 5 seconds to input your extension. Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Medina Sent: Monday, November 07, 2005 11:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Testing with X101P Hi there, im testing my asterisk box using a Modem Intel 56K which on the documentation says it must have the same behavior as an X101P. So im trying to configure just a simple line with 6 extensions. Asterisk loads fine and when im testing an incoming call the welcome message answers but when im trying to dial to any extension, anything happens is like asterisk dont recognice any digit after the welcome. Im using Asterisk version 1.0.9 and im attaching my extensions.conf which is very simple. Any clue will be very helpful. Thanks a lot for your help. Carlos Andres Medina __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] escaping to an extension whilelisteningtovoicemail message
Hope this helps.
exten = s,1,Dial(${ARG1},30,t)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG2})
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(b${ARG2})
exten = s-BUSY,2,Hangup
exten = s-CHANUNAVAIL,1,Voicemail(u${ARG2})
exten = s-CHANUNAVAIL,2,Hangup
exten = s-.,1,Goto(s-NOANSWER,1)
exten = o,1,Hangup
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Monday, November 07, 2005 12:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] escaping to an extension
whilelisteningtovoicemail message
Can you post an example?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Sunday, November 06, 2005 3:32 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] escaping to an extension while
|listeningtovoicemail message
|
|The 'o' works well - especially with the attended transfer function.
|
|PaulH
|
|- Original Message -
|From: Eric ManxPower Wieling [EMAIL PROTECTED]
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|asterisk-users@lists.digium.com
|Sent: Monday, November 07, 2005 4:58 AM
|Subject: Re: [Asterisk-Users] escaping to an extension while
|listening tovoicemail message
|
|
| Anton Krall wrote:
| Guys.
|
| I was wondering, some voicemail systems let you escape to another
|extension
| or context while listening to the voicemail greeting, for
|example, for
| leaving faxes, like Hi, you have reached XXX, if you want
|to leave a
|fax,
| press 5 now, otherwise stay to leave voicemail.
|
| Can this be done on asterisk?
|
| See show application voicemail Pay special attention to the notes
| about the o and a extensions.
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[Asterisk-Users] redial needs the 1 before the prefix howto?
When I go through me call logs I want to redial. The problem is there is no 1 in the number and it won't go through. How do I set it to add the 1 in front of a 10 digit number? -J ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual PRI fail over
On Tue, 11 Oct 2005, Tom wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). Is there any way to set something similar to this up in Asterisk? Tom They are probably talking about NFAS. And after looking at the date of this message, it's probably a moot point. But I'm not sure if NFAS is supported under Libpri/Zaptel. I'd suspect that if it is, it might not be widely deployed and/or tested with it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: * Is it true that if I hand-resolve the server names in all the config files, and then use those instead of the hostnames, this problem won't occur? That's not exactly optimal, of course, since it defeats the whole purpose of dynamic name binding. But it's hard to explain to my SOHO customers, who don't really need any IP-based functionality (although I give all of them some complimentary minutes on nufone) why their phones go down when the Internet is down. Asterisk's use of dns is less then optimal. It expects a response, and if that response contains multiple entries, all entries past the first are ignored. If no response, you already know what happens. Use numeric addresses, etc/hosts, or a dns cache on asterisk. * Is it true that there's no way to get applications in Linux, generally speaking, to try more than a single server when doing a name resolve? Only the first server listed in /etc/resolv.conf (on my systems, anyway) seems to ever get consulted. That's true in most systems. If there are multiple entries, the first one will be attempted, and if no response, the second (etc) will be attempted. If the first responds with _anything_ (even an unknown host), that is considered a response and the dns client will not attempt the second entry in resolv.conf. I think both of these situations are pretty serious failings, if in fact they're failings in the systems and not this bedeviled cranium. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
I was pretty unhappy to see that the new cards had RJ12 sockets - you can put RJ12 into RJ45, but not the other way round... But I do know that a lot of people would ask if RJ12 would fit, so it might have been to cut down on support calls. PaulH - Original Message - From: Kevin Hanson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 1:57 AM Subject: Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance Rich Adamson wrote: Brand new cards, from a recommended Digium distributor. What rev? It'll say Freshmaker Rev x on the blue board. I *think* it was corrected around Rev E or F IIRC, but I don't know for certain. I spoke with Digium and the distributor tech support on it - gave them serial numbers, board rev numbers, etc. No one mentioned this. Probably because your boards are Rev I or later, which I believe are the current boards. FWIW, Rev J is current as of Oct 2005, but it is functionaly equivalent to Rev I. Difference involves some line filtering required by certain country stanadards and use of rj11's instead of rj45 jacks. That's wierd. I have two rev I TDM400's and both have rj11 jacks. As a matter of fact, that caused me a bit of a problem. I was used to rj45. I made up a bunch of patch cables for an install an used cat5 w/rj45. Tried to plug them in and guess what ... did't fit :) Cheers, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
On Sunday 06 November 2005 21:46, [EMAIL PROTECTED] wrote: I was pretty unhappy to see that the new cards had RJ12 sockets - you can put RJ12 into RJ45, but not the other way round... You've gotta be shitting me. Why on earth do you want RJ45 jacks for POTS connections? Sure it fits but it's a loose fit to start and you get absolutely zero advantages unless you count being able to make a screwy cable a good thing. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sangoma a104d
Hi, TE406p seems to be unstable yet. I ordered an sangoma a104d. Does anyone have experience of this card? Thanks. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slightly OT: Firefox search plugin for Voip-info.org
For those using Firefox 1.0+: I have hacked together a search engine install script to enable searching the wiki from the search-bar. http://mycroft.mozdev.org/download.html?name=voipsubmitform=Find+search+plugins If you follow above link and click the search-engine it finds, you'll get an extra engine in the list of search-engines, which allows you to directly invoke the wiki search without having to load voip-info.org first... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie questions
i am pretty new to asterisk. hope to learn more. i have this notice from the console. when i was doing the echo testing by putting the context=default. then, i called out 600 to get the echo test, i can hear the operator talking, but i cant really hear the playback. i am trying to dig around from info from the log files. what does it mean? RFC3389 support incomplete. Turn off on client if possible hope to help..thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Conference-reg
Hi I configured the meetme number in the area where i specified the other extensions but still i am having pbm. herewith i am sending the error i got in the asterisk console. Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 6 19:07:35 ERROR[4952]: chan_zap.c:6731 chandup: Unable to dup channel: No such device or address Nov 6 19:07:35 WARNING[4952]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 6 19:07:35 WARNING[4952]: app_meetme.c:230 build_conf: Unable to open pseudo device regards ramakrishnan.n --- Rich Adamson [EMAIL PROTECTED] wrote: I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 I assume you put the above in meetme.conf file? extension.conf [default] exten = 1234,1,Meetme(1234) Is the [default] section of extensions.conf where all of your other extensions are defined? If not, move the above entry to whatever section you have your other extensions defined. Then stop and restart asterisk. If the above doesn't address your issue, then copy/paste the CLI stuff so we can see what it is telling you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Conference-reg
nr k wrote: Hi I configured the meetme number in the area where i specified the other extensions but still i am having pbm. herewith i am sending the error i got in the asterisk console. Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 6 19:07:35 ERROR[4952]: chan_zap.c:6731 chandup: Unable to dup channel: No such device or address Nov 6 19:07:35 WARNING[4952]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 6 19:07:35 WARNING[4952]: app_meetme.c:230 build_conf: Unable to open pseudo device MeetMe requires that you have a zaptel device (card or emulated via ztdummy). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Conference-reg
If my memory serves you need some kind of zaptel device for meetme to work -- I think even dummy will do, but you need something. on Sunday 11/06/2005 nr k([EMAIL PROTECTED]) wrote Hi I configured the meetme number in the area where i specified the other extensions but still i am having pbm. herewith i am sending the error i got in the asterisk console. Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 6 19:07:35 ERROR[4952]: chan_zap.c:6731 chandup: Unable to dup channel: No such device or address Nov 6 19:07:35 WARNING[4952]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 6 19:07:35 WARNING[4952]: app_meetme.c:230 build_conf: Unable to open pseudo device regards ramakrishnan.n --- Rich Adamson [EMAIL PROTECTED] wrote: I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 I assume you put the above in meetme.conf file? extension.conf [default] exten = 1234,1,Meetme(1234) Is the [default] section of extensions.conf where all of your other extensions are defined? If not, move the above entry to whatever section you have your other extensions defined. Then stop and restart asterisk. If the above doesn't address your issue, then copy/paste the CLI stuff so we can see what it is telling you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] escaping to an extensionwhilelisteningtovoicemail message
Thx! I see now, the o extension has to be on the same context where
voicemail app was called from...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Jennifer Hales
|Sent: Sunday, November 06, 2005 7:25 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] escaping to an
|extensionwhilelisteningtovoicemail message
|
|Hope this helps.
|
|exten = s,1,Dial(${ARG1},30,t)
|exten = s,2,Goto(s-${DIALSTATUS},1)
|exten = s-NOANSWER,1,Voicemail(u${ARG2})
|exten = s-NOANSWER,2,Hangup
|exten = s-BUSY,1,Voicemail(b${ARG2})
|exten = s-BUSY,2,Hangup
|exten = s-CHANUNAVAIL,1,Voicemail(u${ARG2})
|exten = s-CHANUNAVAIL,2,Hangup
|exten = s-.,1,Goto(s-NOANSWER,1)
|
|exten = o,1,Hangup
|
|Regards
|Jenn
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Anton Krall
|Sent: Monday, November 07, 2005 12:16 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] escaping to an extension
|whilelisteningtovoicemail message
|
|Can you post an example?
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of
||[EMAIL PROTECTED]
||Sent: Sunday, November 06, 2005 3:32 PM
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: Re: [Asterisk-Users] escaping to an extension while
||listeningtovoicemail message
||
||The 'o' works well - especially with the attended transfer function.
||
||PaulH
||
||- Original Message -
||From: Eric ManxPower Wieling [EMAIL PROTECTED]
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||asterisk-users@lists.digium.com
||Sent: Monday, November 07, 2005 4:58 AM
||Subject: Re: [Asterisk-Users] escaping to an extension while
||listening tovoicemail message
||
||
|| Anton Krall wrote:
|| Guys.
||
|| I was wondering, some voicemail systems let you escape to another
||extension
|| or context while listening to the voicemail greeting, for
||example, for
|| leaving faxes, like Hi, you have reached XXX, if you want
||to leave a
||fax,
|| press 5 now, otherwise stay to leave voicemail.
||
|| Can this be done on asterisk?
||
|| See show application voicemail Pay special attention to the notes
|| about the o and a extensions.
|| ___
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||http://lists.digium.com/mailman/listinfo/asterisk-users
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|| http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Meetme Conference-reg
Yep,If you do not have a card installed, then you will need to use ztdummy. http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Monday, November 07, 2005 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Meetme Conference-reg If my memory serves you need some kind of zaptel device for meetme to work -- I think even dummy will do, but you need something. on Sunday 11/06/2005 nr k([EMAIL PROTECTED]) wrote Hi I configured the meetme number in the area where i specified the other extensions but still i am having pbm. herewith i am sending the error i got in the asterisk console. Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 6 19:07:35 ERROR[4952]: chan_zap.c:6731 chandup: Unable to dup channel: No such device or address Nov 6 19:07:35 WARNING[4952]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 6 19:07:35 WARNING[4952]: app_meetme.c:230 build_conf: Unable to open pseudo device regards ramakrishnan.n --- Rich Adamson [EMAIL PROTECTED] wrote: I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 I assume you put the above in meetme.conf file? extension.conf [default] exten = 1234,1,Meetme(1234) Is the [default] section of extensions.conf where all of your other extensions are defined? If not, move the above entry to whatever section you have your other extensions defined. Then stop and restart asterisk. If the above doesn't address your issue, then copy/paste the CLI stuff so we can see what it is telling you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Possible Issue With Meetme Conferencing in 1.2.0b2 and latest CVS HEAD (02/11/2005)
Tavis P wrote: Rich Adamson wrote: I'm running Asterisk 1.2.0b2 (also tried latest CVS HEAD) in my lab and i've come across a strange problem. I've setup an extension to call the meetme application, when i call that extension it functions as expected, informing me of my conference number and that i'm the only one in the conference however right after join the conference some problems start occuring: 1. If i call in with another client (both are SIP based), it does not acknowledge the DTMF tones i send to select the conference room, it acts like it never received the DTMF (it plays the please enter the conference number followed by the pound key prompt again) I have verified that the tones are being sent properly, and otherwise work as expected. (before selecting a conference room) 2. When i hang up the phone Asterisk does not clear the SIP channel in use by that phone. Before selecting a conference room calls are properly disconnected by Asterisk and removed from the sip show channels list. 3. After the RTP timeout hits (as configured in sip.conf) it prints a message every second that the call has timed out and will be disconnected. This continues on forever it seems (12 hours in one case) Before selecting a conference room, if left idle (no RTP is sent from SIP UAC), the SIP session is properly disconnected/terminated after the RTP idle timer hits. if add the de options (dynamic, select an empty conference room) the first caller hears the meetme prompts and is put into the first conference room, however the second caller hears nothing, looking at the debug output on asterisk shows that meetme was called and nothing else after that I'm running on linux kernel 2.6.13.4 (vanilla, with grsecurity patches) Zaptel drivers were compiled with make linux26 There is a T100P card in the system and the zaptel and wct1xxp modules are loaded I've tried using the ztdummy module in place of wct1xxp with the same results Asterisk and Zaptel were compiled with gcc 3.3.5 on Debian Sarge submitted bug - http://bugs.digium.com/view.php?id=5578 That's odd. I just checked our meetme using two C7960's and an external Zap (pstn) call, and all worked as expected. Using cvs-head from early morning Nov 1 on fc3 with analog TDM04 card. It seems that this issue is related to my use of the wct1xxp module and a T100P T1 card. After removing the wct1xxp module and loading the ztdummy module in its place the conference bridge works as expected, Asterisk removes (properly) terminated sessions and times out idle sessions. Using the ztdummy module in place of wct1xxp shows a noticable drop in audio quality (blips and phasing) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC errors on PRI
Sounds like a timing issue or interop issue. Get rid of the NFAS (3rd t1 with all B channels) and make them all plain PRIs without D channel sharing. Jason Walker [EMAIL PROTECTED] wrote: I have looked through other postings to the user group for HDLC errors, went through what worked for other people, and still can not seem to get past this issue. For 3 days, I have been getting HDLC abort(6) errors in *. Prior to Tuesday, the circuits were clean...I had maybe 10 HDLC abort messages since August 10th. Here are my specs: 1 Gig IBM x300 w/ 1 Gig Ram 1 Quad TE405P card No errors on IRQs IRQs are separated with NO sharing hdparm for irq and dma are set to 'on' Software - FC1 with -1 updates to kernel, etc. Asterisk v 1.0.9, libpri 1.0.9, zaptel 1.0.9.2 1 T1 is a tieline to our Nortel Meridian 3 T1s are a PRI trunk group with D chans on 24 and 48. The third T1 only has b channels. No alarms from zttool. Calls go through, inbound and outbound. About every 5 seconds, I get the following on the console: Nov 4 21:10:37 NOTICE[9693]: PRI got event: Alarm (4) on Secondary D-channel of span 1Nov 4 21:10:37 NOTICE[9693]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 The errors seem to increase as calls come in and out. There is also a noticable "popping" when the error happens. Any suggestions are welcome. thank you Jason ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! FareChase - Search multiple travel sites in one click. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User language switching in dial plan
Mexican Spanish.. Ha, funny term... :) Mexican Spanish = mx from MeXico... es = ESpain... So, es would be.. humm. Espain Spanish? Chuck Bunn wrote: Hi, What is the best way to allow a user to select the language they hear in the dial plan? In other words I want the phone to answer Hello welcome to ABC company to continue in English press 1 Followed by the same thing in Spanish (Mexican Spanish - I live in the South West United States) but with a press 2. What I would like to avoid is creating two different dial plans and it looks like I can do this, does the following look correct?? By the way is there any prerecorded language selector similar to the above? Or at least something like 'to continue in English press' and 'to continue in Spanish press' the later being in Mexican Spanish. Also I could not find a designator for Mexican Spanish is 'es' correct?? [language] exten = s,1,Answer() exten = s,2,Background(enter-language-extension) exten = 1,1,Set(LANGUAGE()=en) include = internal exten = 2,1,Set(LANGUAGE()=es) include = internal [internal] exten = s,1,Background(enter-ext-of-person) exten = 101,1,Dial(zap/1,10) ... Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme conference pbm using g723.1 codec
Hi all i am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234.I have both sccp ande sip device.if i use the codec ulaw i can able make call between sip and sccp devices and also put meetme conference.if i use g.723.1 codec i have pbm in conference and call between sip and sccp devices.how to solve this pbm.pls do the needful... regards ramakrishnan.n __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
Patching in buildings with rj45 sockets. PaulH - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 2:04 PM Subject: Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance On Sunday 06 November 2005 21:46, [EMAIL PROTECTED] wrote: I was pretty unhappy to see that the new cards had RJ12 sockets - you can put RJ12 into RJ45, but not the other way round... You've gotta be shitting me. Why on earth do you want RJ45 jacks for POTS connections? Sure it fits but it's a loose fit to start and you get absolutely zero advantages unless you count being able to make a screwy cable a good thing. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
Put the ip-address and hostname in /etc/hosts I agree about Asterisk being terrible with DNS failure, but how can you avoid using DNS on *nix system? On 11/7/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Brian Capouch wrote: I don't think this is a new issue--I've seen it talked about on the list before. I don't know if I've ever seen anyone post a fix. My DNS server went out last night in a horrendous storm when an upstream link went down. The madness is that the behavior of the whole server, including the part that's handling my POTS lines, gets wigged out on a DNS failure, making the whole system unusable. I have two questions; being able to solve either would be wonderful: Asterisk is horrible at handleing DNS failures. Don't use DNS with Asterisk. Folkert van Heusden -- Try MultiTail! Multiple windows with logfiles, filtered with regular expressions, colored output, etc. etc. www.vanheusden.com/multitail/ -- Get your PGP/GPG key signed at www.biglumber.com! -- Phone: +31-6-41278122, PGP-key: 1F28D8AE, www.vanheusden.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 3640 as * FXO GW using MGCP?
Hi All,I have searched high and low but cannot find anything on setting up Asterisk using MGCP to talk to a Cisco 3600 FXO gateway.I am running * version 1.0.9 on Slackware 10.I have a Cisco 3640 IOS 12.3.12a IP+ with 2 x NM-2V's and 2 x VIC-2FXO'sUsing SIP to talk between * and the Cisco presents problems with call hang-up for which I was unable to solve. If anyone has info on this I would greatly appreciate it also. If MGCP is possible does anyone have any * and/or Cisco configs?Any help would be greatly appreciated.RegardsMark Cornhill ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to configure adhoc conference in Asterisk
Hi all how to configure adhoc conference in asterisk. regards ramakrishnan.n __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What's the purpose of the username= line?
After some experimentation and posting, I have concluded that in the file sip.conf, the line: username = irrelevant has no effect whatsoever on SIP registration. Any entry on the right hand side will make no difference, and the line can even be avoided altogether (as the O'Reilly book example does). The only effect of that line that have I found is to be displayed under the output of the host*CLI sip show peers command. So, what is that line good for? TIA, -Ramon F Herrera ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
