RE: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup
Just a shot in the dark... but have you tried Answer() before Playback()? Josh McAllister -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Engleward Sent: Monday, May 01, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup I have a PSTN termination provider foo which will accept standard U.S. calls in the form 110 digit ph#. I have an outbound route named foo, with dial pattern 5|., with the only entry in trunk sequence being IAX2/foo. I have an X-lite local extension, on which I can dial 5110 digit ph#, and asterisk will call out over foo and the phone at 10 digit ph# will ring. This rules out a lot of possible problems. extensions.conf includes this: [outgoingtest] exten = s,1,Playback(custom/testmsg) exten = s,2,Wait(1) exten = s,3,Hangup And yes, asterisk has been restarted since the last time any config files were modified. I have a test message at /var/lib/asterisk/sounds/custom/testmsg.gsm If I make the file 1.call containing: Channel: IAX2/foo MaxRetries: 1 RetryTime: 5 WaitTime: 10 Context: outgoingtest Extension: 110 digit ph# Priority: 1 and copy it to /var/spool/asterisk/outgoing/ then the phone doesn't ring, but this shows up on the asterisk console: -- Attempting call on IAX2/foo for 110 digit ph#@outgoingtest:1 (Retry 1) -- Hungup 'IAX2/foo-7' -- Attempting call on IAX2/foo for 110 digit ph#@outgoingtest:1 (Retry 2) -- Hungup 'IAX2/foo-8' The foo-7 and foo-8 on the console are different (numbers anywhere from 1 to 9) every time I try copying the file to outgoing. I tried using extension 5110 digit ph# instead of 110 digit ph# in 1.call, but that didn't work either. Why is it failing? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Error messages
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... well, if you dont use/need a module, in modules.conf put noload = app_intercom.so (for example). i think you can choose whether to automatically load all then specifically noload whichever you dont want with a noload =, or with autoload=no, specify which you want to load. Thank you! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Auto Logout from queue
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... that is a nice function I use a cronjob to logout everyone each evening if anyone wants that script I would love to provide it. Please send the script to the list. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How does asterisk behave when multiple phones are logged in on a single SIP/account?
Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions to this list. Im looking for a way to make the other phones in a group unavailable when one of them is busy. Because one person will have multiple phones. Thanks Arne Morten Johansen. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 482 Loop Detected on sip calls
Hi, Joshua wrote: Why don't you use something like the chan_local channel driver to send the call into the dialplan where it will then execute the extension. Joshua,sorry i don't understand chan_local channel driver to send the call . Could you explain or link to appropriate documentation.Can i do this via the manager interface. You're looping an outbound call back inbound to the same box, with the same callid... so it's perceiving it as a loop. It seems that the UAC (orignating the call) and the UAS (recieving the call) are sharing a transaction database which causes this behaviour.According to RFC 3261 the incoming request are to be matched against server transactions, not client transactions. I am planning on modifying the source to allow the call to be answered rather than detect a loop in this particular case. Also note that the reason i'm doing this is as i need both legs of the call to process FAGI scripts. Regards, Ajit ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 process running concurrent in dialplan
Hi there, I am new to asterisk, I am trying to write a dialplan with 2 process running concurrently. Current dialplan only able to execute process with priority example exten =100,1,Answer()exten =100,2,Musiconhold() exten =100,3,Hangupis it possible to have process musiconhold/background and dial process together.? I am looking into a dialplan where once call connected then the musiconhold or background still running. Mean i need 2 process running at a same time. Is it possible?Thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme volume increase/decrease
Hi. The UPGRADE.txt of asterisk distribution contains the following snippet under the MeetMe heading: "MeetMe: * The conference application now allows users to increase/decrease their speaking volume and listening volume (independently of each other and other users); the 'admin' and 'user' menus have changed, and new sound files are included with this release. However, if a user calling in over a Zaptel channel that does NOT have hardware DTMF detection increases their speaking volume, it is likely they will no longer be able to enter/exit the menu or make any further adjustments, as the software DTMF detector will not be able to recognize the DTMF coming from their device. " My question is... How do you increase/decrease your speaking/listening volume while in a meetme room? Thankxs. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards, so disappointing !
On Sunday 30 April 2006 10:27, Boris Bakchiev wrote: Opened pseudo zap interface, measuring accuracy... This may be a stupid question but how did you do this? -- Cheers Wayne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with SuSe 10
I downloaded the source and built it from that. SuSE10 comes with a version of asterisk 1.0.X on the DVD. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yu Safin Sent: 01 May 2006 16:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with SuSe 10 On 1/24/06, Lee Archer [EMAIL PROTECTED] wrote: Thanks, I've got it running on my test box but didn't know if there was any global objection to using it. I've had a few funnies with it but that might be down to Supermicro and P4's with the EM64T thing. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Klang Sent: 24 January 2006 15:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with SuSe 10 On Tuesday 24 January 2006 09:26, Lee Archer wrote: Has anyone had any experience with the Asterisk on a SuSe 10 platform? I'm currently using FC3 but because we use SuSe within other parts of the business I'm being pushed to changed the OS. Just about all of my production Asterisk servers are on SuSE 9.3. My development and demo boxes are SuSE 10. Both run great. I do however usually tweak the RPM that came with it to add in a few patches. If you are comfortable with running Asterisk 1.0.9 then the RPM works very well. SuSE always seems to really think things through when they package applications. For running something newer than Asterisk 1.0.9 SuSE 10 is also works fine. For your own sanity you'll want to not install/uninstall the SuSE Asterisk RPMs. One possible gotcha: be careful of possibly conflicting kernel modules in /lib/modules/`uname -r`/extra as the Zaptel drivers are not part of any Asterisk package but rather the kernel. The zaptel compile from source installs modules to /lib/modules/`uname -r`/misc so you'll want to delete the files in extra. You'll also have to remember that each time you update the kernel RPM. Hope that helps. The bottom line from me is Thumbs Up. /BAK/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users did you have to install asterisk from source or from rpm? I have installed asterisk under RH and I am switching over to SuSE OSS 10.0. I could not find the rpm for asterisk. My searches show that the rpm is available for the commercial version of SuSE. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme volume increase/decrease
have to use 's' option in MeetMe app, to enable users to go to menu pressing * during conference. When in the menu... well, Allison explains what to do... ;-) 2006/5/2, Jan du Toit [EMAIL PROTECTED]: Hi.The UPGRADE.txt of asterisk distribution contains the following snippet under the MeetMe heading:MeetMe:* The conference application now allows users to increase/decrease their speaking volume and listening volume (independently of each other and other users); the 'admin' and 'user' menus have changed, and new sound files are included with this release. However, if a user calling in over a Zaptel channel that does NOT have hardware DTMF detection increases their speaking volume, it is likely they will no longer be able to enter/exit the menu or make any further adjustments, as the software DTMF detector will not be able to recognize the DTMF coming from their device.My question is... How do you increase/decrease your speaking/listening volume while in a meetme room? Thankxs.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Under which project , auto-dial feature comes
Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random 1-way audio on IAX2 Connections
On 29 Apr 2006, at 03:10, Aryn Nakaoka wrote: I have 2 Asterisk servers connected via IAX2 connections. PBX1 is on the internet with a public IP Address - with PRI PBX 2 is behind a NAT router with IAX2 Ports forwarded 1-way audio is an issue with incoming and outgoing calls using the PRI. However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there are no issues. As well as a restart of asterisk on PBX2 clears up the problem. If you are using IAX2, you don't need to port forward the ports. Just have PBX2 register _often_ and that will keep a mapping in your router. (also you _don't_ need to enable _any_ TCP ports - IAX2 is UDP only) Any bugs in IAX2? If the problem does not go away when you turn off port forwarding, do an iax2 debug on PBX1 and send me a log of a call that fails and I'll take a look. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk with Dialogic BRI /2VFD
Hi all, i have an Asterisk box with an Eicon 4BRI with chan_capi-cm and every thing works fine. We now plan to install a new Asterisk using a Dialogic BRI/2VFD. Is the Dialogic card supported and can i use chan_capi-cm? Has anyone managed to install this card? Unfortunately i was unable to find documentation about Asterisk with Dialogic? thx in advance for your input!!! __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Questions on ANI
I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... Regards, ML ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Armin Schindler wrote: I use two possibilities. a) when I want to connect the ISDN card directly with the device using a short cable, I just cut the cable in the middle an reconnect them crossed and add the resistors here. (Maybe this is not the correct way for termination, but it always worked perfectly. Hi Armin! Thanks for verifying the crossover layout. FYI: http://www.ksi.at/online-kataloge/kat8A/8A-109/8A-109.htm I've ordered some of them - I do not like soldering :-) regards klaus b) when there is a ISDN bus (cables and plug wall mounted), I just add the cross and terminations to these boxes on the wall. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Questions on ANI
If the incoming call is from a PRI with DIDs, you could just simply map the caller ID name to the company name (company name being the name of the company being called, this is opposite of normal caller id name) based on the DID. Or, you could also play a sound file containing the company name. I'm looking at doing that with queues. We have multiple businesses all sharing the same office/employees/etc, and I plan to have a queue for each company. When the receptionist picks up the phone, she will hear an announcement with the name of the company that the person called. On 5/2/06, Li Mark [EMAIL PROTECTED] wrote: I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... Regards, ML___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using frequent keepalives to eliminate need for NAT port forwarding?
On 2 May 2006, at 01:25, Tom Engleward wrote: I have an asterisk system behind NAT, and need to connect to public PSTN originators via SIP or IAX2, but don't have the option of forwarding any ports (4569, 5060, etc) to the asterisk system. However, the NAT system does properly establish transient UDP forwarding on the basis of outgoing connections, so is it possible to configure asterisk to send frequent keepalive UDP packets (say every 30 seconds) from ports 4569 and 5060 to the PSTN originators in order to keep the NAT system's transient forwarding in effect, so that when the PSTN originator receives inbound calls and attempts to contact my asterisk system, the NAT system won't drop the packets? Yes. That is the way that IAX2 likes to work. However, not all providers will allow it, some require a fixed IPaddress and port for them to send calls to. My home Asterisk is behind a NAT router and is set up like that. add something like this to iax.conf: register = username:[EMAIL PROTECTED] By the way, In case anyone is interested, I've got an NSLU2 running asterisk 1.0.10 quite nicely. No fan, no moving parts, less that £100 total outlay. Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetAsterisk London and Brussels
Just a quick reminder - now is the last chance to register for MeetAsterisk in Brussels on Thursday and London on Friday. We have updated the web site with location information and will keep registration open until tomorrow lunch. http://www.meetasterisk.com See you! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling
On a full cone NAT, I have never been able to get the ATA to register without a stun. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, May 02, 2006 1:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling My experience is that a stun server does not do anything that nat=yes in asterisk does not do. Asterisk is capable of determining the source port and ip address of a registration, so there is no need for the UA (ATA) to learn this information form a stun server. Keep it simple if possible, the stun server just adds another device to manage and/or worry about being unreachable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, May 01, 2006 9:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling You will probably want to set a stun server in the 2100 if behind a nat. You can use stun.fwdnet.net for testing. With that, you probably wont need to port forward it should work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, May 01, 2006 8:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling Set nat=yes as you have Enable qualify=yes Important - Do a sip reload or asterisk reload (the nat and qualify settings have to be refreshed, at least with realtime and rtcahcefriends). Turn off all NAT traversal features on the SPA2100 If it still does not work - your NAT router may be the issue, make sure that security policy allows ALL outbound traffic from the SPA2100 (no filters). With Linksys, Belkin, and some 3com/USR NAT routers (among others I am sure) you will need to make sure you have recent firmware on them, older firmware (1 year or older in many cases) does not behave well with SIP and NAT. The NAT=yes tells asterisk to use the IP address and port of the connection socket (a form of NAT discovery similar to a STUN server), not what is in the registration message, and the qualify=yes tells asterisk to send periodic SIP OPTIONS queries to keep the NAT timeout from expiring on the NAT router. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Lyons Sent: Monday, May 01, 2006 5:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 1 - 3 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see it get an OK and sip show peers shows it on the list. But I can't get a dial tone. It works fine connecting to a local Asterisk box (not traversing NAT). Anyone know the magic trick? My sip.conf looks like: [homesip] type=friend username=homesip secret=pw context=fagi ;qualify=yes host=dynamic nat=yes tried qualify both ways. My sip show peers says: telebox*CLI sip show peers Name/username HostDyn Nat ACL Port Status homesip/homesip67.188.35.109D N 5060 Unmonitored Can't seem to find enough info to get this to work, any help appreciated greatly, Eric. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by
[Asterisk-Users] Zapata Telephony interface and torisa module error
Looking at my log fileI found the following error: May 2 12:00:45debian kernel: Zapata Telephony Interface Registered on major 196May 2 12:00:45 debian kernel: No ISA tormenta card found at dMay 2 12:00:45 debian kernel: Zapata Telephony Interface UnloadedMay 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output errorMay 2 12:00:45 debian insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesgMay 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: insmod char-major-196 failed Isthe torisa modulethe problem? ..end Zapata Telephony Interface, is it the torisa module? Giuseppe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to set outgoing callerid
Hi, just answering myself: I am not allowed to send the leading 0 for my prefix with the callid, then it works well. Sebastian Sebastian Reitenbach [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Tue, 2 May 2006, Klaus Darilion wrote: Armin Schindler wrote: I use two possibilities. a) when I want to connect the ISDN card directly with the device using a short cable, I just cut the cable in the middle an reconnect them crossed and add the resistors here. (Maybe this is not the correct way for termination, but it always worked perfectly. Hi Armin! Thanks for verifying the crossover layout. FYI: http://www.ksi.at/online-kataloge/kat8A/8A-109/8A-109.htm I've ordered some of them - I do not like soldering :-) Thanks for the link. But these devices do not seem to have a crossed connection. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue Transfer
HiDinesh, thanks for help me. To activate the transferences of calls in asterisk, I effected: SIP.CONF in sip of the agent I qualified canreinvite=no, so that asterisk monitors this transference. EXTENSIONS.CONF I qualified the parameters tT in the command Dial FEATURES.CONF I qualified [ featuremap ] to blindxfer = # ; to atxfer = * 7 Greetings Josué 2006/5/1, Dinesh Nair [EMAIL PROTECTED]: On 04/29/06 20:15 Josué Conti said the following: Dinesh the agents they receive a call and this call will have to be transferred, them uses only functions hold and trnsf in devicei'm not sure how the polycom's hold and trnsf buttons are mapped, but usingblindxfer and atxfer dtmf keypresses marks the agent unused upon a transfer. --Regards, /\_/\ All dogs go to heaven.[EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+| for a in past present future; do|| for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b.|| done; done|+=+ ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Extreme delay before * processes call files
Found it! It seems that Asterisk is looking at the date / time stamp of the call file to process the call?? I was simply moving the call files hoping it would just work (tm) I guess that the call files created on the samba share I created carried the time/date stamp of the local machine (workstation) and not the asterisk server causing a time difference. Now I run a touch * on the asterisk server before moving the call files, all the calls are now processed immediately. Is this intended behaviour for the call files?? Or just a bug? Thanks On Fri, 28 Apr 2006, Remco Barende wrote: I guess that I'm the only one experiencing this problem is there any way to debug this problem? Does anyone know how to debug this particular item in *? (Or should I open a bug in Mantis?) Thanks!! On Thu, 27 Apr 2006, Remco Barende wrote: Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call file just keeps sitting in the /var/spool/asterisk/outgoing directory and it seems that * is doing nothing with it?? Only after 10-30 seconds sometimes even much longer the call file is picked up. There is no message on the * console about a call file being present. Does anyone have a clue why asterisk fails to pick up call files within a reasonable amount of time? The load on the box is 0.05 at most. Thanks!! Remco ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi...Please help me
Hi friends,Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and "X-Lite" as softphone in my PC and client PC. Here my user name is "chandra" and client user name is "aarti". I have added these lines to configuration files at the end of file.added contents in sip.conf:[aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial[chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorialadded contents in extensions.conf:[tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra)Here, "aarti" is client, "chandra" is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using "X-Lite" softphone. But, when i try to connect from "aarti" system using softphone, it displays an error message "login timedout, contact system admin". Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you.Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk "clients" are phones not PCs unless you use"soft phones" which is software onthe PC(somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo.The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably oughtto take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asteriskand do some reading before you even start as it will help you fit many pieces of the asterisk "puzzle" together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Extreme delay before * processes call files
On Tue, 2006-05-02 at 14:52 +0200, Remco Barende wrote: Found it! It seems that Asterisk is looking at the date / time stamp of the call file to process the call?? I was simply moving the call files hoping it would just work (tm) I guess that the call files created on the samba share I created carried the time/date stamp of the local machine (workstation) and not the asterisk server causing a time difference. Shows why accurate time keeping is important in a network environment. Sort out the offending client would be my choice. Is this intended behaviour for the call files?? How else would wake up calls work? Or just a bug? Seems you answered it yourself. On Fri, 28 Apr 2006, Remco Barende wrote: I guess that I'm the only one experiencing this problem is there any way to debug this problem? Does anyone know how to debug this particular item in *? (Or should I open a bug in Mantis?) -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi...Please help me
can u check what this command gives iptables -L or do iptables -F [ Not advisable , but for testing OK ] then try again --- Crazy Boy [EMAIL PROTECTED] wrote: Hi friends, Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and X-Lite as softphone in my PC and client PC. Here my user name is chandra and client user name is aarti. I have added these lines to configuration files at the end of file. added contents in sip.conf: [aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial [chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorial added contents in extensions.conf: [tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra) Here, aarti is client, chandra is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using X-Lite softphone. But, when i try to connect from aarti system using softphone, it displays an error message login timedout, contact system admin. Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you. Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk clients are phones not PCs unless you use soft phones which is software on the PC (somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably oughtto take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asterisk and do some reading before you even start as it will help you fit many pieces of the asterisk puzzle together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula - Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Send instant messages to your online friends http://uk.messenger.yahoo.com ___ Win tickets to the 2006 FIFA World Cup Germany with Yahoo! Messenger. http://advision.webevents.yahoo.com/fifaworldcup_uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi...Please help me
You are missing the dtmf mode, and most importantly the codec to be used. I would also add the nat=yes, that is probably why your phone isnt registering. See below for example config: [chandra] type=friend username=chandra secret=chandra nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 context=tutorial canreinvite=no From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Crazy Boy Sent: Tuesday, May 02, 2006 8:58 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Hi...Please help me Hi friends, Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and X-Lite as softphone in my PC and client PC. Here my user name is chandra and client user name is aarti. I have added these lines to configuration files at the end of file. added contents in sip.conf: [aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial [chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorial added contents in extensions.conf: [tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra) Here, aarti is client, chandra is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using X-Lite softphone. But, when i try to connect from aarti system using softphone, it displays an error message login timedout, contact system admin. Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you. Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk clients are phones not PCs unless you usesoft phones which is software onthe PC(somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably ought to take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asteriskand do some reading before you even start as it will help you fit many pieces of the asterisk puzzle together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards, so disappointing !
On 5/2/06, Wayne Gemmell [EMAIL PROTECTED] wrote: Opened pseudo zap interface, measuring accuracy... This may be a stupid question but how did you do this? in your zaptel source dir (after making..): ./zttest -v or search for zttest on voip-info. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP trunk ring tone
Hi, I'm wondering what I need to change to get the swedish type ring on a SIP-trunk. When I make an inbound call i still have the US-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Using qualify=yes guarantees failure on iax2 behind NAT (was: RE: [Asterisk-Users] Using frequent keepalives to eliminate need forNAT port forwarding?)
--- Damon Estep [EMAIL PROTECTED] wrote: Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open, if you use 1 to 1 NAT (versus PAT where it is many to one NAT) it will work because port 5060 on the private address will still be port 5060 on the public address. Tried that, and it just turned an intermittent failure into a permanent failure. I added: qualify=yes qualifyfreqnotok=15000 qualifyfreqok=2 qualifysmoothing=yes to the peer details for the iax trunk in [EMAIL PROTECTED] and hit the big red reload line at the top. Then iax2 show peers on the console showed under status that the peer was indeed being monitored and was ok and had a ping of about 100ms, and iax2 debug showed all the keepalive messages every 20 seconds, as intended. And calling to my assigned DID using my PSTN provider's own outbound termination (so that the call was both outbound and inbound on my iax2 trunk), the call worked as usual. But calling from an external phone (so that my iax2 trunk would see only the inbound connection), my asterisk system failed to ever answer at all, and iax2 debug showed no indication that it ever even noticed any incoming call. So I deleted those four qualify lines and hit [EMAIL PROTECTED]'s big red reload, yet iax2 show peers STILL showed the peer being monitored! And asterisk still refused to answer external incoming calls. So I did restart gracefully and asterisk finally actually honored my deletion of the qualify lines (sip show peers now once again shows status as Unmonitored, as before), and once again asterisk notices and answers incoming calls placed not only from my PSTN provider's own termination but also from external phones... though of course it's probably going to start failing intermittently again, as usual. So now I have a new question (besides my original, about how to ensure that asterisk _always_ answers the phone): why would enabling qualify cause an immediate and consistent failure to ever answer incoming external phone calls? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Armin Schindler wrote: Thanks for verifying the crossover layout. FYI: http://www.ksi.at/online-kataloge/kat8A/8A-109/8A-109.htm I've ordered some of them - I do not like soldering :-) Thanks for the link. But these devices do not seem to have a crossed connection. No, just termination. klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Configuration
Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten = 19,1,Dial(SIP/19,20,tr) exten = 19,2,Voicemail(u19) exten = 19,hangup exten = 19,102, Voicemail (b19) exten = 19,103,Hangup exten = 20,1,Dial(SIP/20,20,tr) exten = 20,2,Voicemail(u20) exten = 20,hangup exten = 20,102, Voicemail (b20) exten = 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Configuration
my guess is that you are trying to dial a sip channel to reach an iax peer. Dial(SIP/19) should be Dial(IAX2/19) Olivier Saulnier wrote: Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten = 19,1,Dial(SIP/19,20,tr) exten = 19,2,Voicemail(u19) exten = 19,hangup exten = 19,102, Voicemail (b19) exten = 19,103,Hangup exten = 20,1,Dial(SIP/20,20,tr) exten = 20,2,Voicemail(u20) exten = 20,hangup exten = 20,102, Voicemail (b20) exten = 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using frequent keepalives to eliminate need for NAT port forwarding?
--- Tim Panton [EMAIL PROTECTED] wrote: Yes. That is the way that IAX2 likes to work. Ok. However, not all providers will allow it, some require a fixed IPaddress and port for them to send calls to. Is this the reason for the recommendation I've seen in various forums to have port 4569 forwarded to the asterisk machine? Do either of the providers (teliax, exgn) I've seen recommended elsewhere on this list require a fixed port to send calls to? A possibly related issue: At http://www.teliax.com/forum/viewtopic.php?t=173sid=bb1196132c2eee0ca4a0e09bd04d5309 in a conversation which took place in December of 2005, somebody wrote: Do a iax2 show registry from the CLI, and you will notice what ports Teliax 'thinks' you are at... it should not be :4569 since your NAT router has picked a different return port for you. But for me, the entry for perceived is always my NAT router's public IP:4569. Somebody else in that forum said the same is true for them too, and there was a very interesting reply: I've recently uncovered a periodic problem in the NAT kernel module in linux. It effects both 2.4 and 2.6 linux kernels. The problem is when the source port and destination port are the same on a UDP connection (IAX2 is exactly that). If you sniff the traffic comming out of your router you will find that when Asterisk can't register the source address in the outgoing packets are still your private IP address behind the router. This is why it never gets pasted the first packet going out. Teliax's asterisk box will send the reply back to a non existant IP address. This effects most routers on the market since embeded Linux is the most common OS for these. Perhaps this is the source of my (and probably some other people's on this mailing list too) current frustration. Perhaps due the particular nature of this Linux bug and the fact that PSTN origination/termination providers normally use port 4569 on their own machines for IAX2, a suitable workaround for an asterisk machine behind a buggy NAT router would be to simply use some local UDP port other than 4569 for IAX2 connections? How do you configure asterisk to use a nonstandard local port for IAX2? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Configuration
Yes Sean, I've just see that :-) I modify, and my communication is now OK. But i always have the message on the console... Bets regards, OLS Sean Cook a écrit : my guess is that you are trying to dial a sip channel to reach an iax peer. Dial(SIP/19) should be Dial(IAX2/19) Olivier Saulnier wrote: Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten = 19,1,Dial(SIP/19,20,tr) exten = 19,2,Voicemail(u19) exten = 19,hangup exten = 19,102, Voicemail (b19) exten = 19,103,Hangup exten = 20,1,Dial(SIP/20,20,tr) exten = 20,2,Voicemail(u20) exten = 20,hangup exten = 20,102, Voicemail (b20) exten = 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup
--- Josh McAllister [EMAIL PROTECTED] wrote: Just a shot in the dark... but have you tried Answer() before Playback()? http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+answer-before-playback says New versions of Asterisk have added Answer capabilities to several functions like Playback(), which means that those functions will answer themselves if necessary. That new versions is as of January 2005, and I'm running [EMAIL PROTECTED] 2.8, released April 2006. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help configuring TE100P and 3 X100P clone with MD3200 chipset
I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# What's wrong with configuration? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata Telephony interface and torisa module error
Giuseppe Parlato wrote: Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: insmod char-major-196 failed Is the torisa module the problem? ..end Zapata Telephony Interface, is it the torisa module? Giuseppe Giuseppe, You probably just don't have the original Tormenta ISA card. You can ignore this message. Better yet, configure your system to not load the torisa.o module. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dnd error message in the log
Is this a problem? What is dnd anyway?Thanks,Jim.May 2 10:44:08 DEBUG[6277] db.c: Unable to find key 'SIP/201' in family 'dnd' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random 1-way audio on IAX2 Connections
On 2 May 2006, at 15:32, Tom Engleward wrote: --- Tim Panton [EMAIL PROTECTED] wrote: If you are using IAX2, you don't need to port forward the ports. Just have PBX2 register _often_ and that will keep a mapping in your router. Where is this set? Is this the minregexpire and maxregexpire settings in iax.conf, which default to 60 seconds? (And what's the purpose of the min and max; why isn't there just one setting, simply regexpire?) And is the default of 60 seconds considered often? Finally, what's the difference between registering often and setting qualify=yes? 'Often' is often enough for your Nat router. 60 secs seems enough for the ones I've used. If I recall it right: Qualify sends an IAX POKE, (expecting a PONG) whereas registration sends a full REGREQ which requires more processing power at the far end (typically a database lookup of your user name and an MD5 of your password) and an extra 2 packets to complete the exchange. You are 'encouraged' to qualify, but I like to register, mostly out of habit I suppose. Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset
Are you seriously trying to run 4 cards in one system? The odds of getting that working are about the odds of Angelina Jolie showing up on my doorstep ready to whisk me off tobut I digress...you will have serious interrupt issues trying to get 4 cardss working in one system. I am surprised that you would fork for a PRI card but use cheap winmodems for analog lines. You will have much better luck tossing the x100p cards and using either SPA-3000's, a TDM400, or a Mediatrix 1204. Kerry Garrison Publisher - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Tuesday, May 02, 2006 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# What's wrong with configuration? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does asterisk behave when multiple phones are logged in on a single SIP/account?
The last sip device to register gets the call. The way around this is to have your sip devices register under different accounts and create a ring group (dial(SIP/dev1SIP/dev2SIP/devN))AFAIK, there isn't a reliable method of determining if a sip device is busy other than calling it. On 5/1/06, Arne Morten Johansen [EMAIL PROTECTED] wrote: Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions to this list. I'm looking for a way to make the other phones in a group unavailable when one of them is busy. Because one person will have multiple phones. Thanks Arne Morten Johansen. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help configuring TE100P and 3 X100P clone with MD3200 chipset
On Tue, 2 May 2006, Wai Wu wrote: [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# What's wrong with configuration? What's wrong is that configuration in your /etc/modprobe.conf is automatically running ztcfg when you load wcte11xp. The ztcfg fails because it is expecting all the card drivers to be loaded. So remove all the install lines from your /etc/modprobe.conf, then run ztcfg -vv manually once you have loaded all the drivers. Or, just ignore the error on loading wcte11xp and plow on loading wctdm. At that point the ztcfg should succeed. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dnd error message in the log
Is this a problem? What is dnd anyway? Not a problem, probably dialparties.agi checking if this extension as DND enabled. DND stand for Do Not Disturb hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help configuring TE100P and 3 X100P clone with MD3200 chipset
Hi Wai Wu, is seeming that you have two problems. In the case of wcte11xp, I find that its board TE110P is unplugged of slot PCI, removes the board and restarts the server. Later it again board the TE110P in slot correctly and restarts the server, must resolv this problem. In the case of wcfxo, I find that this problem is in relation to the modules, if will be able effects one again make clean and one make and make install of asterisk. I wait to have helped. Regards 2006/5/2, Wai Wu [EMAIL PROTECTED]: I can either get the TE100P working or the 3 X100P clones working, butnever both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company.This is my zaptel.confspan=1,1,0,d4,amifxsks=1-24loadzone=usfxols=25-27loadzone=usI then do[EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobewcte11xpZT_CHANCONFIG failed on channel 25: No such device or address (6)/lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed/lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxoZT_CHANCONFIG failed on channel 25: Invalid argument (22)Did you forget that FXS interfaces are configured with FXO signallingand that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed/lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed[EMAIL PROTECTED] root]#What's wrong with configuration?___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata Telephony interface and torisa moduleerror
You probably just don't have the original Tormenta ISA card. You can ignore this message. Better yet, configure your system to not load the torisa.o module. first thanks.. I think so, I guess I don't have it. However I even don't know what it is and why it was needed to be configured. I'm not sure that this module is really loaded cause lsmod doesn't show it. I can find it in /etc/modules.cgf : options torisa base=0xd alias char-major-196 torisa post-install tor2 /sbin/ztcfg post-install torisa /sbin/ztcfg ... After I got this log message this morning I had to restart ser just some minutes later, don't you think there is some reletionship log message and problem with ser? giuseppe - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 02, 2006 4:43 PM Subject: Re: [Asterisk-Users] Zapata Telephony interface and torisa moduleerror Giuseppe Parlato wrote: Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: insmod char-major-196 failed Is the torisa module the problem? ..end Zapata Telephony Interface, is it the torisa module? Giuseppe Giuseppe, You probably just don't have the original Tormenta ISA card. You can ignore this message. Better yet, configure your system to not load the torisa.o module. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.5.1/328 - Release Date: 01/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dnd error message in the log
DND = do not disturb. Sounds like you are running [EMAIL PROTECTED] The error below just means that on an incoming call, it looked to see if you set a flag for DND in the database. You didnt, so it just went to the next number in the dial plan. That is what is supposed to happen. bp From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jim Lynch Sent: Tuesday, May 02, 2006 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] dnd error message in the log Is this a problem? What is dnd anyway? Thanks, Jim. May 2 10:44:08 DEBUG[6277] db.c: Unable to find key 'SIP/201' in family 'dnd' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling zaptel
Hello, When I compile zaptel application, i have this error log file: cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -lm gendigits.c -o gendigits ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file ZAPTELVERSION=SVN-trunk-r980 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp gcc -I/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/drivers/net/wan -I/include/net -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from zaptel.c:43: /usr/include/linux/kernel.h:72: error: erreur de syntaxe before size_t /usr/include/linux/kernel.h:74: error: erreur de syntaxe before size_t In file included from /usr/include/linux/timex.h:186, from /usr/include/linux/sched.h:11, from /usr/include/linux/module.h:10, from zaptel.c:45: /usr/include/linux/time.h:14: error: erreur de syntaxe before time_t /usr/include/linux/time.h:16: error: erreur de syntaxe before '}' token /usr/include/linux/time.h:20: error: erreur de syntaxe before time_t In file included from /usr/include/linux/timex.h:186, from /usr/include/linux/sched.h:11, from /usr/include/linux/module.h:10, from zaptel.c:45: /usr/include/linux/time.h: Dans la fonction « timespec_to_jiffies »: /usr/include/linux/time.h:198: error: dereferencing pointer to incomplete type /usr/include/linux/time.h:199: error: dereferencing pointer to incomplete type /usr/include/linux/time.h: Dans la fonction « jiffies_to_timespec »: /usr/include/linux/time.h:219: error: dereferencing pointer to incomplete type I cut here, because the end of file is in error too :-) I use the latest release of zaptel. The problem is that the type size_t in /usr/include/linux/kernel.h is not recognised. If i replace it by int type, the compilation is OK until i have the type time_t in time.h file :-( Di you have any idea for that (how to do accept this types, or remplace them by what?? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata Telephony interface and torisa moduleerror
You probably just don't have the original Tormenta ISA card. You can ignore this message. Better yet, configure your system to not load the torisa.o module. first thanks.. I think so, I guess I don't have it. However I even don't know what it is and why it was needed to be configured. I'm not sure that this module is really loaded cause lsmod doesn't show it. I can find it in /etc/modules.cgf : options torisa base=0xd alias char-major-196 torisa post-install tor2 /sbin/ztcfg post-install torisa /sbin/ztcfg ... After I got this log message this morning I had to restart ser just some minutes later, don't you think there is some reletionship log message and problem with ser? giuseppe - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 02, 2006 4:43 PM Subject: Re: [Asterisk-Users] Zapata Telephony interface and torisa moduleerror Giuseppe Parlato wrote: Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: insmod char-major-196 failed Is the torisa module the problem? ..end Zapata Telephony Interface, is it the torisa module? Giuseppe Giuseppe, You probably just don't have the original Tormenta ISA card. You can ignore this message. Better yet, configure your system to not load the torisa.o module. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.5.1/328 - Release Date: 01/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip show inuse
I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use sip show inuse. Below is the output... I know there are current calls: redhat*CLI sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Commands possible in the h extension, message delivery with confirmation
Hi, I've been using Asterisk for several months now with great success. I'm working on a system that tries to deliver a recorded message to a user, as follows: 1. a call file is placed in /var/lib/asterisk/outgoingcalls 2. This triggers a call to be placed 3. When answered, the caller hears the recorded message 4. After the message, the caller must confirm by pressing 1 5. If they confirm, the process ends 6. If they just hang up, wait 7. After waiting, dial them again and go to step 3 above The idea is that the system persists in trying to deliver the message, not quitting until they press '1' on their keypad after hearing the recorded message. I plan to extend this to try several different users in rotation until confirmation is received. I need to re-dial the user if they hang up, or time out, or are busy. For hang-up, I have tried using the h extension, like this: h,1,Dial(local/201) To test re-dialling on hangup. However, the dial takes place too soon after the previous call, resulting in an error from the target phone. When I change it to: h,1,Wait(3) h,2,Dial(local/201) This gets no further than the Wait command, and * never executes the Dial command. Can this be made to work reliably? Or should I write some sort of AGI script to provide a reliable message delivery application to do this instead? I'm using Asterisk 1.2.4 on Centos 4. Thanks, Jeremy Tucker Network Manager regs4ships Ltd. Tel: +44 (0)870 444 1240 Fax: +44 (0)23 8022 8029 Email: [EMAIL PROTECTED] Web: www.regs4ships.com Skype: jez_r4s CONFIDENTIALITY NOTICE The contents of this email are confidential to the ordinary user of this email address to which it was addressed and may also be privileged. If you are not the addressee of this email you may not copy, forward, disclose or otherwise use any part of it in any form whatsoever. If you have received this email in error please email the sender by replying to this message. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speeding up UK BT incoming call detection
Hi, I am running Asterisk v1.2.7.1 with a Digium TE110P. My dialplan is very simple, when a call comes in on my analogue BT PSTN line, it rings the other ZAP interface (my house phone). Slightly pointless (having a 1x1 switch) I know, but I am planning on doing more with internal SIP extensions, and outgoing SIP services etc At the moment it's just simple though whilst I try to fix this problem. My problem is that it normally takes 2-3 rings on an incoming call before Asterisk detects the line ringing and then a further 2 rings before it starts ringing my phone. I know this because I have plugged a normal analogue phone into another extension of the BT line as my Asterisk connected phone starts riniging about 4 or so rings later! Can anyone tell me what the optimal settings for zaptel.conf and zapata.conf for a BT line or have any other suggestions to try and speed things up? Cheers Richard Dutton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need help configuring TE100P and 3 X100Pclonewith MD3200 chipset
This is a system for our lab. I have no problem getting rid of X100P clones. But I am just curious why can they work. Even the drivers are not loading correctly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, May 02, 2006 10:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Need help configuring TE100P and 3 X100Pclonewith MD3200 chipset Are you seriously trying to run 4 cards in one system? The odds of getting that working are about the odds of Angelina Jolie showing up on my doorstep ready to whisk me off tobut I digress...you will have serious interrupt issues trying to get 4 cardss working in one system. I am surprised that you would fork for a PRI card but use cheap winmodems for analog lines. You will have much better luck tossing the x100p cards and using either SPA-3000's, a TDM400, or a Mediatrix 1204. Kerry Garrison Publisher - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Tuesday, May 02, 2006 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# What's wrong with configuration? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky
I have dial through application, that uses the wW options on the dial command. However it's seems to be really hit or miss if asterisk picks up the *1 and starts the recording. It can take 3 or 4 attempts before I can see from the console that's it's started recording. A user just on the call not able to see the console has no chance of knowing if it was started or not. does anyone know of any other area's of the system that I can tweak that would make this a bit less flakey. Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Questions on ANI
May I know if you could send me some coding on the *.conf, so that I can follow the idea that you suggest?? ML 2006/5/2, Lacy Moore - Aspendora [EMAIL PROTECTED]: If the incoming call is from a PRI with DIDs, you could just simply map the caller ID name to the company name (company name being the name of the company being called, this is opposite of normal caller id name) based on the DID. Or, you could also play a sound file containing the company name. I'm looking at doing that with queues. We have multiple businesses all sharing the same office/employees/etc, and I plan to have a queue for each company. When the receptionist picks up the phone, she will hear an announcement with the name of the company that the person called. On 5/2/06, Li Mark [EMAIL PROTECTED] wrote: I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... Regards, ML ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset
Hmm, I don't have /etc/modprobe.conf, and wctdm is giving me problems. Which device is it talking about? [EMAIL PROTECTED] /]# modprobe wctdm /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wctdm.o: insmod /lib/modules/2.4.20-8/misc/wctdm.o failed /lib/modules/2.4.20-8/misc/wctdm.o: insmod wctdm failed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 02, 2006 10:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Need help configuring TE100P and 3 X100P clonewith MD3200 chipset On Tue, 2 May 2006, Wai Wu wrote: [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed /lib/modules/2.4.20-8/misc/wcte11xp.o: insmod wcte11xp failed [EMAIL PROTECTED] root]# modprobe wcfxo ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# What's wrong with configuration? What's wrong is that configuration in your /etc/modprobe.conf is automatically running ztcfg when you load wcte11xp. The ztcfg fails because it is expecting all the card drivers to be loaded. So remove all the install lines from your /etc/modprobe.conf, then run ztcfg -vv manually once you have loaded all the drivers. Or, just ignore the error on loading wcte11xp and plow on loading wctdm. At that point the ztcfg should succeed. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip show inuse
I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Piper Sent: Tuesday, May 02, 2006 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sip show inuse I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use sip show inuse. Below is the output... I know there are current calls: redhat*CLI sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as a phone survey system
Grab your fav. bottle of
${Insert_Your_Fav_Booz_bottle_brand_here} and get
working on it.
--- TV JOE [EMAIL PROTECTED] wrote:
I write perl applications for a living and have
developed code to
talk to all kinds of hardware. What I'd like to do
is pull a list of
phone numbers from sql via dbi and call each. An
initial voice
messsage would be played asking the recipient if
they'd
optionally like to fill out our survey. If so I'd
like to on the
fly play pre-recorded questions and record the
touch tone
response back into the database. Teleyapper looks
like it
does some of what I want but I'm not sure slicing
it up is
better than starting from scratch. There appear to
be a few
CPAN modules to work with Asterisk. I'm looking for
advice
on how hard this is to implement with Asterisk.
TIA, TV JOE
Kerry Garrison [EMAIL PROTECTED] wrote:
Asterisk is simply a telephony toolkit, so the
simple answer is yes, Asterisk can do this. Also,
being a toolkit means there are a number of ways to
accomplish it. You could right PERL, Python, TCL, C,
PHP or numerous other types of scripts that can
manage this for you. To see how to do some of the
basic functions, you can look at some of the scripts
at Nerd Vittles (http://nerdvittles.com). Things
like the TeleYapper will give you a basis to work
from.
Kerry Garrison
Publisher - http://GeekGazette.com -
http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of TVJOE
Sent: Wednesday, April 26, 2006 7:31 PM
To:asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asteriskas a phone
survey system
Hi,
I'm interested in developing anautomated phone
survey and am curious
if Asterisk could beconfigured to run such a
system.. My idea is to
record a message anda series of sub-questions.
The system would
call each number on alist and play the message,
Depending on the
touch tone responseanother message would be
played. Is it possible
for asterisk tomanage a survey like this? If so
can the responses from
thelisteners be recorded. If someone else has
done this I'd beinterested
in details.
TIA , TV JOE
-
Yahoo!Messenger with Voice. PC-to-Phone calls
for ridiculously low
rates.___
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-
How low will we go? Check out Yahoo! Messengers low
PC-to-Phone call rates.
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RE: [Asterisk-Users] Need help configuring TE100P and 3 X100Pclonewith MD3200 chipset
Are you seriously trying to run 4 cards in one system? The odds of getting that working are about the odds of Angelina Jolie showing up on my doorstep ready to whisk me off tobut I digress...you will have serious interrupt issues trying to get 4 cardss working in one system. I am surprised that you would fork for a PRI card but use cheap winmodems for analog lines. You will have much better luck tossing the x100p cards and using either SPA- 3000's, a TDM400, or a Mediatrix 1204. Kerry Garrison Publisher - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com I agree with Kerry about the insane Interrupt load on the machine, Go for a TDM400 and toss the clones. I gotta go Brad and AJ are waiting on the HeliPad... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip show inuse
Not sure if this helps any, but I had no clue what sip show inuse would show until I found out you had to put incominglimit (or call-limit) in sip.conf/realtime for it to know that the phone was in use... Not sure if this'll help. On Tue, 2 May 2006, Alexander Lopez wrote: I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Piper Sent: Tuesday, May 02, 2006 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sip show inuse I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use sip show inuse. Below is the output... I know there are current calls: redhat*CLI sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telasip config problem/question
I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air.I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.cominsecure=veryqualify=yestype=usercontext=from-pstnThen I went to incoming routes and set it up for any did, any cid, but the telasip connection is taking a different route. Here are the log entries for both. Telasip:May 2 11:11:55 DEBUG[2670] chan_sip.c: Checking SIP call limits for device jlynchMay 2 11:11:55 DEBUG[2670] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]May 2 11:11:55 VERBOSE[6343] logger.c: -- Executing Set(SIP/jlynch-cf63, FROM_DID=6782280738) in new stackMay 2 11:11:55 VERBOSE[6343] logger.c: -- Executing Answer(SIP/jlynch-cf63, ) in new stack May 2 11:11:55 VERBOSE[6343] logger.c: -- Executing PlayTones(SIP/jlynch-cf63, ring) in new stackMay 2 11:11:55 DEBUG[6343] channel.c: Scheduling timer at 160 sample intervalsMay 2 11:11:55 VERBOSE[6343] logger.c: -- Executing NVFaxDetect(SIP/jlynch-cf63, 0) in new stackMay 2 11:11:55 DEBUG[6343] app_nv_faxdetect.c: Preparing detect of fax (waitdur=4ms, sildur=1000ms, mindur=100ms, maxdur=-1ms) May 2 11:11:55 DEBUG[2670] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match FoundMay 2 11:11:55 DEBUG[2670] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED]' of Response 103: Match FoundMay 2 11:11:55 DEBUG[6343] channel.c: Scheduling timer at 0 sample intervals May 2 11:11:55 DEBUG[6343] app_nv_faxdetect.c: Got hangupZap:May 2 10:55:09 DEBUG[2670] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]May 2 10:55:09 VERBOSE[6287] logger.c: -- SIP/200-9b14 answered Zap/1-1May 2 10:55:09 DEBUG[6287] chan_zap.c: Requested indication -1 on channel Zap/1-1May 2 10:55:09 DEBUG[6287] channel.c: Scheduling timer at 0 sample intervalsMay 2 10:55:15 DEBUG[6287] channel.c: Didn't get a frame from channel: SIP/200-9b14May 2 10:55:15 DEBUG[6287] channel.c: Bridge stops bridging channels Zap/1-1 and SIP/200-9b14May 2 10:55:15 DEBUG[6287] chan_sip.c: update_call_counter(200) - decrement call limit counter May 2 10:55:15 DEBUG[6287] app_dial.c: Exiting with DIALSTATUS=ANSWER.May 2 10:55:15 VERBOSE[6287] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/1-1' in macro 'dial'May 2 10:55:15 VERBOSE[6287] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/1-1' in macro 'exten-vm'May 2 10:55:15 VERBOSE[6287] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/1-1'I even attempted to add an incoming route using the Telasip did, but it didn't work either. I have the radio button clicked in each of them to route the call to extension 200. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with Dialogic BRI /2VFD
richard Coco wrote: Hi all, i have an Asterisk box with an Eicon 4BRI with chan_capi-cm and every thing works fine. We now plan to install a new Asterisk using a Dialogic BRI/2VFD. Is the Dialogic card supported and can i use chan_capi-cm? Has anyone managed to install this card? Unfortunately i was unable to find documentation about Asterisk with Dialogic? thx in advance for your input!!! Richard, i think the only dialogic cards that with work are the jct models. then i think you need to buy drivers from digium hope this helps Tom -- This message has been scanned for viruses and dangerous content and is believed to be clean. Thank You For Choosing Cache Communications ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as a phone survey system
But code quickly, as the quality produced is inversly related to the
amount of ${Insert_Your_Fav_Booz_bottle_brand_here} in your system.
Grab your fav. bottle of
${Insert_Your_Fav_Booz_bottle_brand_here} and get
working on it.
--- TV JOE [EMAIL PROTECTED] wrote:
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RE: [Asterisk-Users] Speeding up UK BT incoming call detection
Isn't a TE110P a PRI card? Are you sure that's the right model number for an analogue interface card? For our sites with BT lines, we have them configured as follows (I've extracted the settings I think might be relevant): usecallerid=no hidecallerid=no callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes I think disabling asterisk from getting caller ID off an analogue line improves its answering speed considerably. Of course, if you want CLID info off your analogue line (and are paying BT for the privilege), you may not wish to go down this route. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip show inuse
It does work now... thanks. annoyed I guess I need to add that line to all sip users if I want to monitor who is on the phone and who isn't. /annoyed Thanks again, bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, May 02, 2006 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Sip show inuse Not sure if this helps any, but I had no clue what sip show inuse would show until I found out you had to put incominglimit (or call-limit) in sip.conf/realtime for it to know that the phone was in use... Not sure if this'll help. On Tue, 2 May 2006, Alexander Lopez wrote: I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Piper Sent: Tuesday, May 02, 2006 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sip show inuse I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use sip show inuse. Below is the output... I know there are current calls: redhat*CLI sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1443 (20060314) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip show inuse
Actually, I take that back... It still isn't working. It does show the users peers now, but they stay at 0. I set this on our SIP carrier made both an incoming and outbound call... it still showed 0 during the call. Any other ideas? Thanks, bp -Original Message- From: William Piper [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 02, 2006 12:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Sip show inuse It does work now... thanks. annoyed I guess I need to add that line to all sip users if I want to monitor who is on the phone and who isn't. /annoyed Thanks again, bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, May 02, 2006 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Sip show inuse Not sure if this helps any, but I had no clue what sip show inuse would show until I found out you had to put incominglimit (or call-limit) in sip.conf/realtime for it to know that the phone was in use... Not sure if this'll help. On Tue, 2 May 2006, Alexander Lopez wrote: I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Piper Sent: Tuesday, May 02, 2006 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sip show inuse I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use sip show inuse. Below is the output... I know there are current calls: redhat*CLI sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip show inuse
Not to mention the obvious, and this may not help your situation, but if you were (or are) using templates it would be a one-line change. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Piper Sent: Tuesday, May 02, 2006 12:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Sip show inuse It does work now... thanks. annoyed I guess I need to add that line to all sip users if I want to monitor who is on the phone and who isn't. /annoyed Thanks again, bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, May 02, 2006 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Sip show inuse Not sure if this helps any, but I had no clue what sip show inuse would show until I found out you had to put incominglimit (or call-limit) in sip.conf/realtime for it to know that the phone was in use... Not sure if this'll help. On Tue, 2 May 2006, Alexander Lopez wrote: I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Piper Sent: Tuesday, May 02, 2006 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sip show inuse I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use sip show inuse. Below is the output... I know there are current calls: redhat*CLI sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1443 (20060314) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speeding up UK BT incoming call detection
On Tue, May 02, 2006 at 05:39:53PM +0100, Chris Bagnall wrote: [snip] I think disabling asterisk from getting caller ID off an analogue line improves its answering speed considerably. Of course, if you want CLID info off your analogue line (and are paying BT for the privilege), you may not wish to go down this route. With an analogue line CallerID is presented b/m the 1st and 2nd ring (which means waiting for 2 rings). Also with Digium hardware to produce CallerID you have to tell it to do two rings or you get internal s/w timeouts (Emailed them about it and they never replied). With UK PRI it should be sent with the call set-up in the data channel, not the bearers. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: MWI on Treo 600/650
Andrew Kohlsmith wrote: On Thursday 13 April 2006 09:02, David Cook wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. I've been working on this off and on for AGES. There are some SMS portal sites that claim to be able to do this as well, but I have not managed to find one. -A. I know this thread is probably a little aged, but I'm intrigued... How are you forwarding cell vm to asterisk? When busy or unavailable, do you forward to a DID set up to go directly to your asterisk voicemail? I get so many complaints about how the buttons to navigate Asterisk voicemail are different from the company's cell phones and different again from their personal cell phones. I could combine at least two of them this way! Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] billing realtime
How about AstRTB ? Asterisk Real Time Billing --- Thameem Ansari [EMAIL PROTECTED] wrote: Hello All, I had the same question when I was writing my own billing software in java. Here is what I am doing to track multiple calls at a time from the prepaid account. 1. Keep on db table for balance and reserver_balance. 2. First call coming to agi, check the balance - Sum of all the reserve_balance of that account code. 3. Check the destination and allowed minutes for that balance amount from step 2. 4. Reserve balance table will contain destination, amount, reserved secs columns 5. If the avaialable balance is = 0 then announce not enough credit and hangup. 6. If the available balance is 0 but seconds allowed to talk is less than reserved secs (see step 8 for more details about what this is) then set absolutetimeout for those seconds. 7. Otherwise the allowed seconds is more than the allowed seconds, set absolute time out for the reserved seconds and make the call. 8. Reserved secs is a custom constant seconds, say you can reserve fund for 3 minutes (180 seconds). if the account has balance for only 2 minutes (120 seconds) then the absolute time out will be 120 seconds. 9. Once the channel status changed to reserved, insert an record to reserve_balance table with uniqueid, accountcode, amount, reserved_secs information. The above steps will handle one call so far now...and lets see how the dial plan should be, 10. In your dial plan, add an AbsoluteTimeout extension T and call another AGI script which will just to reset the absolute timeout. 11. When the particular timeout is reached asterisk will transfer the call to 'T extension which will in turn call another agi. 12. The agi will receive all the information about the channel including uniqueid, repeat the steps 2- 7 (except dial) and reset the abstimeout and this process will repeat until the channel hangup. 13. Once the channel hangup, you can either use Manager to receive the cdr event or you can set h extension (not reliable and not recommended) to calculate the real balance and update the balance table. Once you update the balance table, remove the record from reserve_balance table for the uniqueid, channel and accountcode. (these three are enough to find out the entry in that table). Now lets take the scenario for second call when the first call was active, 14. When the second call comes in, start from step 2. In step 2, we are doing finalBalance = Balance - Sum(reserve_balance) for that account code. If there is already a call on this accountcode, then this table will have one entry and the reserved amount. Get the finalBalance by subtracting the amounts. Follow step 3 and allow or deny the caller. The above said solution is very stable and doesn't overflow the memory or session and not using any threads. The only restriction here is, if we have the scenario, Call -1 balance = $0.10 destination= 1 (which is US) rate = $0.02 per minute reserveSecs = 10 minutes (600secs) finalBalance = $0.10 - $0 (consider this is first call and no entry in reserve_balance table) = $0.10 allowedMints = $0.10/$0.02 = 5 minutes = 300 seconds. AbsoluteTimeout = 300 seconds (this is less than the default reserveSecs so set this as abstimeout) Call -2 balance = $0.10 destination= 1 (which is US) rate = $0.02 per minute reserveSecs = 10 minutes (600secs) finalBalance = $0.10 - $0.10 (consider this is second call and already an entry in reserve_balance table) = $0.0 allowedMints = 0 seconds. announce the denied ivr. So, the reserveSecs is critical to avoid how much threshold amount the caller should have to make two calls. If they have $10 in their account as per the above two algorithms, they can make as many simultaneous calls. I hope this solves most of your problems. I looked at ASTCC, A2Billing etc and they are not doing this way and not know whether they work properly. But this works for me. Shoot me your questions if you have one. I am developing my own billing and routing app (in java) and I need a name for that.. guys pls suggest one.. i may put that in sourceforge if i feel confident. Thanks, Thameem On 4/27/06, JP Carballo [EMAIL PROTECTED] wrote: Dovid Bender wrote: A while back some one posted some code that he used that took out the flag in astcc that kept track if there was a call in progress for that pin or not. Dont know if it wil work for real time though. Dovid I don't know if you were pertaining to what I posted in the message ASTCC: How to reset in-use flag automatically ?. The setinuse() routine already exists in ASTCC. One simply has to use that routine to disable the inuse flag when a call begins and ASTCC will allow multiple calls for the same account. However, I too have no idea if this will work for real time. -- JP Carballo
Re: [Asterisk-Users] Asterisk as a phone survey system
How about the Call Progress Analysis ?
Mike
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 02, 2006 11:14 AM
Subject: RE: [Asterisk-Users] Asterisk as a phone survey system
Grab your fav. bottle of
${Insert_Your_Fav_Booz_bottle_brand_here} and get
working on it.
--- TV JOE [EMAIL PROTECTED] wrote:
I write perl applications for a living and have
developed code to
talk to all kinds of hardware. What I'd like to do
is pull a list of
phone numbers from sql via dbi and call each. An
initial voice
messsage would be played asking the recipient if
they'd
optionally like to fill out our survey. If so I'd
like to on the
fly play pre-recorded questions and record the
touch tone
response back into the database. Teleyapper looks
like it
does some of what I want but I'm not sure slicing
it up is
better than starting from scratch. There appear to
be a few
CPAN modules to work with Asterisk. I'm looking for
advice
on how hard this is to implement with Asterisk.
TIA, TV JOE
Kerry Garrison [EMAIL PROTECTED] wrote:
Asterisk is simply a telephony toolkit, so the
simple answer is yes, Asterisk can do this. Also,
being a toolkit means there are a number of ways to
accomplish it. You could right PERL, Python, TCL, C,
PHP or numerous other types of scripts that can
manage this for you. To see how to do some of the
basic functions, you can look at some of the scripts
at Nerd Vittles (http://nerdvittles.com). Things
like the TeleYapper will give you a basis to work
from.
Kerry Garrison
Publisher - http://GeekGazette.com -
http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of TVJOE
Sent: Wednesday, April 26, 2006 7:31 PM
To:asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asteriskas a phone
survey system
Hi,
I'm interested in developing anautomated phone
survey and am curious
if Asterisk could beconfigured to run such a
system.. My idea is to
record a message anda series of sub-questions.
The system would
call each number on alist and play the message,
Depending on the
touch tone responseanother message would be
played. Is it possible
for asterisk tomanage a survey like this? If so
can the responses from
thelisteners be recorded. If someone else has
done this I'd beinterested
in details.
TIA , TV JOE
-
Yahoo!Messenger with Voice. PC-to-Phone calls
for ridiculously low
rates.___
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How low will we go? Check out Yahoo! Messenger's low
PC-to-Phone call rates.
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Re: [Asterisk-Users] OT: MWI on Treo 600/650
On 4/13/06, David Cook [EMAIL PROTECTED] wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. I've had this working using Clickatell.com as a third-party provider, and by using Gammu (gammu.org) with a Nokia 6190 (using serial connection). MWI is set by setting the UDH of the SMS to '04010200xx' where the last two digits indicate the number of new messages -- when 00, the MWI is turned off, other values should display on the phone. BTW, not all cel-carriers allow you to change the message-center number for voicemail -- they've disabled the menu option in the phones to prevent you from changing it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Using qualify=yes guarantees failure on iax2 behind NAT (was: RE:[Asterisk-Users] Using frequent keepalives to eliminate needforNAT port forwarding?)
So now I have a new question (besides my original, about how to ensure that asterisk _always_ answers the phone): why would enabling qualify cause an immediate and consistent failure to ever answer incoming external phone calls? Because the firmware on your NAT router has an unconditional timeout on NAT entries, so the qualify = yes can not keep the NAT open. Have you tried upgrading your router firmware and or trying a different router? We use qualify and nat exclusively with SIP. Not IAX, so if there is an IAX related unique issue I would not have any experience. We have hundreds of SIP devices behind NAT routers with no NAT traversal other than qualify and nat = yes and the only problems we have ever had have been related to old and/or junky routers, most of which can be fixed by upgrading the firmware or replacing the router with a basic consumer router from a reputable manufacturer. The endpoints are all of the Linksys/Sipura variety or Polycom phones, none of the NAT traversal features in the device are enabled. I should also add that our asterisk servers are NOT behind any NAT, only the SIP UAs are. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: MWI on Treo 600/650
On Tuesday 02 May 2006 13:10, Mark Johnson wrote:
I know this thread is probably a little aged, but I'm intrigued... How
are you forwarding cell vm to asterisk? When busy or unavailable, do
you forward to a DID set up to go directly to your asterisk voicemail?
heh... I cheat. I don't give out my cellphone number. I give out a DID going
to Asterisk, and then do this:
exten = 5551212,1,Dial(${ANDREWCELL},16,rt)
exten = 5551212,n,VoiceMail([EMAIL PROTECTED])
exten = 5551212,n,Macro(handle-hangup)
Note that this is one of the ONLY times that it is *RIGHT* to use the 'r' Dial
option, by the way. I use it because if my cell is off/dead/out of area the
caller hears ringing instead of The cellular customer you have called is not
available.
I have call waiting on the cell, so that feature continues to work as expected
and, if I don't answer, it just goes to voicemail.
I get so many complaints about how the buttons to navigate Asterisk
voicemail are different from the company's cell phones and different
again from their personal cell phones. I could combine at least two of
them this way!
There was some mention of all the key sequences being programmable at one
point for the Voicemail app... I haven't heard anything about that in a while
though.
-A.
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Re: [Asterisk-Users] OT: MWI on Treo 600/650
Mark Johnson wrote: Andrew Kohlsmith wrote: On Thursday 13 April 2006 09:02, David Cook wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. I've been working on this off and on for AGES. There are some SMS portal sites that claim to be able to do this as well, but I have not managed to find one. -A. I know this thread is probably a little aged, but I'm intrigued... How are you forwarding cell vm to asterisk? When busy or unavailable, do you forward to a DID set up to go directly to your asterisk voicemail? I get so many complaints about how the buttons to navigate Asterisk voicemail are different from the company's cell phones and different again from their personal cell phones. I could combine at least two of them this way! I do the same thing. It's called conditional-call-forward or call-divert, depending on what continent you find yourself on. Under normal circumstances, CCF is set to the cellular provider's voicemail system, which in turn uses called# info to put you into the right voice-mail. Since I don't have a PRI which provides that kind of info, I simply use a DID for it. Nice feature on T-Mobile US is that you have a 500 minute bucket of CCF minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The CAVP is now accepting memberships applications
On-line signup form are available on our website at www.cavp.ca in the Membership section or please call 1-866-940-CAVP (2287) and select option 3 (CAVP treasurer). -- The CAVP is now accepting memberships applications. This is a pivotal moment for the CAVP and we need your support. Establishing a solid membership base will determine the organizations ability to function as an effective association. CAVP Membership is available to any interested party who is involved in the Voice Over IP industry. Further details including a link to our on-line signup form are available on our website at www.cavp.ca in the Membership section. Founding members will be given special recognition within the CAVP and on the CAVP web site including prominent links to your business. To be considered a founder, members must join soon and also pay a one time membership fee which is twice their normal yearly fee. Benefits for Members Membership in CAVP provides many benefits which can have a direct impact on your profitability. CAVP Members: - Have the right to display the CAVP logo on the web sites and any of their business material. - Are eligible to vote at CAVP meetings for board elections and general policy directions. - Have representation at CRTC working group meetings for issues such as E911, ENUM, and network interconnection. - Can ask the CAVP for legal advice with regard to regulatory issues and compliance. - Granted access to members only area of the web site containing complete listings of members contact information as well as documents as listed below. - Allowed to receive and post to the CAVP mailing lists. - Can utilize the CAVP's library of legal documents and disclaimers for customers to ensure they are in compliance with CRTC regulations. - Entitled to use the CAVP's ENUM system for direct peering with other CAVP members. - Participation in the CAVP IP Gateway service which allows members to terminate calls through other members PSTN gateways in each region. Disclaimer: As this organization is newly formed not all of the member benefits are fully in place at this time. -- Regards, John Lange Canadian Association of VoIP Providers 1-866-940-CAVP (2287) ext 2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dial 'R' option gone?
In article [EMAIL PROTECTED], Benoit Panizzon [EMAIL PROTECTED] wrote: On Friday 28 April 2006 15:32, Eric ManxPower Wieling wrote: What does the R option do? Indicate 'Ringing' as soon as the called party indicates 'Ringing'. The 'r' option indicates 'Ringing' as soon as the connection is built, even if the called party is not yet ringing. With some SIP Services I then had the situation that the call was 'hanging' on the gateway, gut the caller heard a ringing, so tought the called phone would be ringing which was not the case. Apparently R also works with 1.2.5 even if not documented. Are you certain the version in which you had the 'R' option was standard Asterisk? Could it have been a BRIstuff version instead? Or with some other patch? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRIs from two different telco
On Apr 28, 2006, at 8:45 AM, Don Pobanz wrote: Wai Wu wrote: One question thought, does the hardware echo cancellation work much better than software? I bought a Digium TE411P hoping the hardware echo canceler would improve my echo problems over the software echo canceler, but had no performance improvement. Since then I have heard that the Digium T1 cards use the same algorithm for both hardware and software echo canceling so hardware will only work better if your CPU is overloaded when doing software echo canceling. I don't know where you heard this, but the hardware echo can and the software can are completely different, bearing no sort of common heritage. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing extensions in a call group.
Hi all, I've got an Asterisk at home system running the new Free PBX front. It's solved all our small office VOIP phone system which we are using as our only source of telephone communications. Anyway,I have set up a few ring groups. The first rings the internal office extensions. After 15 seconds it switches to the second ring group which rings two cell (mobile) numbers and should ring the same extensions as in the first ring group Ring group 1: extension 100, extension 101 Ring group 2: group 200, group 201, extension 100, extension 101 Ring group 200: cell number A Ring group 201: cell number B All groups are set to ringall When an incoming call arrives, extension 100 rings and after a delay of 5 seconds or so, extension 101 joins in. 10 seconds later both extensions stop and cell number A starts ringing. The problem is that Cell number B or extensions 100 and 101do not ring. I want 200, 201, Extension 100 101 all to ring together while on ring group 2. Why does asterisk not do this? Kind regards Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk technician needed in Buenos Aires Argentina
Dear guys:We are expanding our voip unit and currently looking for an Asterisk technician that can be part of our company here in Buenos Aires. If you know anyone who lives here and knows Linux and Asterisk, please contact me asap. Best regards,Sergio Veltriwww.pointhorizon.comSuipacha 119 Primer pisoCapital FederalBuenos Aires, Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with multiple company setup
Hello Everyone, Here is the scenario... I have a client who has two different companies in the same officebut everyone works for both companies. Each person has a DID for both companies. They only want to have one phone at their desk. They have purchased the GXP-2000 ip phones for the project. I would like to setup account 1 for the first company and account 3 for the second company inside the gxp config. For my intitial testing I have setup two phones. The first phone is registered to both companies in the example I described above. The second phone is just registered for one company. Now from the phone with both accounts created I can dial out to the other phone from both accounts and it shows the correct company when dailing. But I am unable to dial the second company from the other phone (ext 200). I can dial the first company (ext 100) just fine. When dialing ext 200 I am receiving 484 messages. Any ideas? - Jason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma Card Question
Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Transfer Disconnect
Hey everyone, I have a TE110P card hooked up to a PRI and about 40 Grandstream GXP 2000 phones using it. Wheneverwe transfer an incoming call using the builtin GXP2000 transfer button or using the Asterisk blind transfer, the caller is disconnectedif theextension is busy.Is this how blind transfers are supposed to work?I am trying to find a way to ensure that if the extension is not picked up the caller still goes to voicemail. I am using asterisk 1.2.4 PRI portion of my zaptel.conf: span=1,1,0,esf,b8zsbchan=1-23dchan=24 PRI portion of my zapata.conf: --- switchtype=nationalcontext=from-pstnsignalling=pri_cpegroup=10channel = 1-23 As a side question: Would it be possible to create a blind transfer application using the application map in features.conf? Thanks in advance, John Coleman [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi...Please help me
First off, I agree w/ Gonzalo – softphones didn't work out for me either. One thing that did work great tho was a combo. We share SIP phones at the office in a 1:4 ratio. You're probably asking – how do you know when a ringing phone is for you? Well, everyone in our office gets an XLite softphone, and I direct calls to make BOTH the SIP phone AND the XLite ring. If your XLite pops up, you know that ring phone is for you… Here's some answers to your other questions • What I have to install in client PC's? Just the softphone client (e.g. XLite (SIP) Cubix (IAX) http://www.virbiage.com/cubix.php • What hardware I need? Nothing too fancy. Your PCs seem OK. For Asterisk, I'm using an old Pentium 4 beater with 1Gig memory and it handles the whole office (19) just fine. • How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server) This depends on what you want in the way of handsets, and what kind of connectivity you want to the PSTN (Public Switched Telephone Network). You could get away with no extra hardware in a pure VoIP solution. Connect Asterisk to the Internet w/ an Ethernet cable and use SIP based phones that also communicate over a network. Note that if you don't use any Digium hardware, I believe that you need to use ztdummy to control timing (never used it myself) http://www.voip-info.org/wiki-Asterisk+timer+ztdummy • Is it sufficient to buy hardware for server only OR for client PC's also? Again, your PCs seem OK. How you kit out your server depends upon what you want. • How can I connect my VoIP phone to server? Once you have Asterisk installed, you have to configure your VoIP phone to register with it. For example, look here for how to configure Polycom Soundpoint 501s - http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501. You'll also have to have the appropriate entries in SIP.conf for the phone AND to connect to your VoIP service provider http://www.voip-info.org/wiki-Asterisk+config+sip.conf • How can I connect hardware to server? Don't understand this one. If you use telephony boards, you'll need drivers. Depending upon the board you may also have to physically connect your phone to it with a telephone wire (as is the case with TDM boards for example) • How can I connect PSTN line to server PC? Assuming analogue phones you'll need a TDM card with an FXO port (outgoing) for each line you have (http://www.digium.com/en/products/hardware/analogcards.php). You'll also need an FXS port for each phone you have on your TDM card as well. Yours, H On 5/2/06, William Piper [EMAIL PROTECTED] wrote: You are missing the dtmf mode, and most importantly… the codec to be used. I would also add the nat=yes, that is probably why your phone isn't registering. See below for example config: [chandra] type=friend username=chandra secret=chandra nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 context=tutorial canreinvite=no From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Tuesday, May 02, 2006 8:58 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Hi...Please help me Hi friends, Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and X-Lite as softphone in my PC and client PC. Here my user name is chandra and client user name is aarti. I have added these lines to configuration files at the end of file. added contents in sip.conf: [aarti] type=friend username=aarti secret=aarti host=dynamic context=tutorial [chandra] type=friend username=chandra secret=chandra host=dynamic context=tutorial added contents in extensions.conf: [tutorial] exten = 101,1,Dial(SIP/aarti) exten = 102,1,Dial(SIP/chandra) Here, aarti is client, chandra is mine and Asterisk is also installed in my PC (chandra) and it is successfully connected to Asterisk server using X-Lite softphone. But, when i try to connect from aarti system using softphone, it displays an error message login timedout, contact system admin. Is there any problem with the content of sip.conf file or extensions.conf file? I have not connected any external hardware to my pc. I just want to connect Asterisk server to my collegues PC's like Intercom within my office LAN using headphones. How can I do this? Please tell me. Looking forward for your response. Thank you. Regards, Chandra. Evalyn Wafula [EMAIL PROTECTED] wrote: Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk clients are phones not PCs unless you use soft phones which is software on the PC (somewhat like Skype) that you use to
Re: [Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO
Mike Clark [EMAIL PROTECTED] wrote: If you aren't going for the echo cancellation, then I think either card will do fine. We are now deploying only the A200 because we never know if echo will be an issue or if it can be tuned away Thanks, Mike. That's a good point in favour of the A200 - it's cheap to add the hardware echo canceller if it's needed. Nick. -- Nick Chalk . once a Radio Designer Confidence is failing to understand the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO
John Novack [EMAIL PROTECTED] wrote: Though many will probably disagree, you will be LOTS better off with the Sangoma A200 It is MUCH more forgiving regarding Motherboards and the PCI 2.2 requirement, That's one of my concerns. I'm working with refurbished hardware, so don't have much freedom of choice in motherboards. I'm planning on using a fairly good Intel dual- socket-370 server board, but I can't afford to make the wrong choice. even though the software installation instructions leave much to be desired for the inexperienced, Is that inexperienced in telephony, or inex- perienced in Linux? I have zero telephony experience, but plenty with Linux. :-) Nick. -- Nick Chalk . once a Radio Designer Confidence is failing to understand the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO
Chris Bagnall [EMAIL PROTECTED] wrote: The site in Northampton with 3 FXO has been an absolute nightmare over the last 9 months the system's been in place. Once asterisk 1.2 was deployed, things improved remarkably. Do you think that was improved code in v1.2, or the result of your calibration? I took the time on-site to properly configure the txgain and rxgain using a milliwatt tone coming from one of our other servers in a datacentre Is that procedure documented somewhere? I've found the asterisk config for generating the tone, but what are you measuring on the test system to set the correct gains? One thing worth checking with BT - if you can find someone who can give you an accurate answer :-) - is whether the Featureline will give you disconnect supervision. If it will, so much the better, as * 1.2 seems to have usable support for it. Thanks, I'll look into that. It's still not as good as the building next door where they have 2 ISDN BRIs Yes, that might make life easier. I don't think I can make the argument for converting to ISDN yet, though. :-/ Thanks, Chris! Nick. -- Nick Chalk . once a Radio Designer Confidence is failing to understand the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi...Please help me
On Tuesday 02 May 2006 16:42, hugolivude wrote: We share SIP phones at the office in a 1:4 ratio. You're probably asking – how do you know when a ringing phone is for you? Well, everyone in our office gets an XLite softphone, and I direct calls to make BOTH the SIP phone AND the XLite ring. If your XLite pops up, you know that ring phone is for you… That seems to be humongous overkill... why not just use any of the caller ID popup apps instead of running that behemoth X-Lite? If the popup comes up, the phone's for you. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardware
I am not by any means recommending this to anyone but I wanted to publish this for reference. I have an Asterisk system connected to a provider via IAX trunks. There are 32 phones on our network and we have about 400 calls per day to/from our system. The hardware running this is a Pentium Pro 400mhz with 256MB ram and a 9GB scsi hard drive. Everything is working great even on such meager hardware. Our other systems are Dual Xeon servers with 1 or 2GB of ram each handling our PRI's and customer systems. -Jonathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Half hangup issue
All, I have this issue happening on 2 seperate asterisk boxes, it happen from version 1.2.4 i am currently running version 1.2.7. What happens is i will be on a call, and all of a sudden I will hear a fast busy, the person that i was talking to can still hear me fine. It doesn't really matter how long the call is, sometimes its 10 minutes in, sometimes its 2 hours in. I have a mix of grandstream products i am using, a gxp2000, and bt102, and a handy 386. It really only seems to happen on the handy, but not exclusively. Has anyone else seen this behavior? I am at a loss, my provider first thought it might be something with IAX so we switched our protocol to sip, and it lessened the problem, but it still occurs. I have not been able to track the problem down, as * still thinks the call is fine even though its obviously not. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma Card Question
Maybe some kind of callwaiting/threewaycalling activated on that? The system is identifying the hang up as a flash. -Original Message- From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 2 May 2006 16:30:56 -0400 Delivered: Tue, 02 May 2006 17:28:33 Subject:[Asterisk-Users] Sangoma Card Question Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1146602038.579235.10961.baladonia.terra.com.br,4024,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky
When I hit *1 in my system, I got a beep to let me know that the recording started. Is this not happenning to you? []'s MM -Original Message- From: Mark Ackroyd [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Cc: Sent: Tue, 2 May 2006 16:39:10 +0100 Delivered: Tue, 02 May 2006 12:36:09 Subject:[Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky I have dial through application, that uses the wW options on the dial command. However it's seems to be really hit or miss if asterisk picks up the *1 and starts the recording. It can take 3 or 4 attempts before I can see from the console that's it's started recording. A user just on the call not able to see the console has no chance of knowing if it was started or not. does anyone know of any other area's of the system that I can tweak that would make this a bit less flakey. Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1146585728.278418.29759.baladonia.terra.com.br,3947,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Hi...Please help me
I guess that would work to if I knew about any caller-id popup apps! Wasn't that much overkill actually, we all had XLite installed for our failed soft-phone trial. Besides some users travel and take the XLites w/ them... Anyway the idea's the same and that's what's important. Howard On 5/2/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 02 May 2006 16:42, hugolivude wrote: We share SIP phones at the office in a 1:4 ratio. You're probably asking – how do you know when a ringing phone is for you? Well, everyone in our office gets an XLite softphone, and I direct calls to make BOTH the SIP phone AND the XLite ring. If your XLite pops up, you know that ring phone is for you… That seems to be humongous overkill... why not just use any of the caller ID popup apps instead of running that behemoth X-Lite? If the popup comes up, the phone's for you. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP trunk ring tone
Setting the country=se in [general] context inside indications.conf didn't work? []'s MM -Original Message- From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: Sent: Tue, 2 May 2006 15:32:24 +0200 Delivered: Tue, 02 May 2006 07:36:20 Subject:[Asterisk-Users] SIP trunk ring tone Hi, I'm wondering what I need to change to get the swedish type ring on a SIP-trunk. When I make an inbound call i still have the US-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1146576980.726213.11352.balcomo.terra.com.br,3682,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec G729 no longer works.
I just tested this out and I am working again. Thanks for the great advice. Thanks Again -Jason On Sun, 2006-04-30 at 19:27 +0200, Mathieu Chouquet-Stringer wrote: [EMAIL PROTECTED] (Patrick) writes: Looks like an SELinux issue. Try booting with selinux=0 or disable SELinux in /etc/sysconfig/selinux, reboot and see if it works then. If you to double check it is a SELinux issue, no need to reboot: 'setenforce permissive' will (temporarily) do the trick (man setenforce for more information) -- Jason A. Kates ([EMAIL PROTECTED]) Fax:208-975-1514 Phone: 212-400-1670 x2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Insights on SIP channel usage in * 1.2.7.1 are welcome!
I've had a heck of a time getting a SIP channel to work in Asterisk
1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on
pre 1.2 versions of Asterisk so perhaps it's version related. Any
insights are welcome!
At first I wasn't able to dial out on the SIP channel _whenever_ I
started Asterisk (i.e. not just when the box was booted). I always
had to do a reload from the CLI before it would work. Using Ethereal
I noticed that there seemed to be some trouble resolving my ITSP's
hostname sip.unlimitel.ca (althogh I cannot explain why it would
_always_ start working after a reload) so I ended replacing the
hostname sip.unlimitel.ca with the actual IP address (64.26.157.251).
Not pretty but at least I can call out now. BTW adding:
sip.unlimitel.ca64.26.157.251
to hosts didn't help.
I'd be grateful for any insights on this and whether there's a more
elegant sol'n.
Anyway I was able to call out on that SIP channel but I couldn't
receive calls on it. I captured a SIP debug trace and noticed
something about the SIP number not being in the context. The context
associated w/ the SIP channel looked like this:
[incoming]
exten = s,1,NoOp(${CONTEXT})
exten = s,n,Ringing()
exten = s,n,GoTo(attendant-MainMenu,s,1)
exten = s,n,Hangup()
I found that I had to add:
exten = _6477235412,1,NoOp(${CONTEXT})
exten = _6477235412,n,Ringing()
exten = _6477235412,n,GoTo(attendant-MainMenu,s,1)
exten = _6477235412,n,Hangup()
I found this odd because I thought s would be sufficient (it has been
in the past). Any comments you can share w/ me on this?
I've also noticed this warning message from time-to-time in the CLI:
WARNING[2203]: chan_sip.c:9633 handle_response_register: Got 200 OK on
REGISTER that isn't a register
Any ideas?
My SIP.conf is below. BTW what's auth=md5 supposed to do. I can't
find any documentation on it so I commented it out.
Many Thanks,
H
; ---
; /etc/asterisk/sip.conf
;
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding nat=1 to each peer definition to
; solve translation problems.
;
;
[general]
;
context=incoming-bogus-calls
bindport=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine)
maxexpirey=3600 ; Must be larger than the re-register timeout
on the router
defaultexpirey=3600
notifymimetype=text/plain
rtptimeout=60
rtpholdtimeout=300
disallow=all
allow=ulaw
;
; This section is because i'm behind nat
;
;register=6477235412:mysecret@sip.unlimitel.ca/6477235412
register=6477235412:mysecret@64.26.157.251/6477235412
externip=mystaticIPaddress ;Outside address
localnet=192.168.0.148/255.255.255.0 ;Inside Network
;
;
[6477235412]
type=peer
;auth=md5
username=6477235412
fromuser=6477235412
fromdomain=unlimitel.ca
secret=mysecret
;host=sip.unlimitel.ca
host=64.26.157.251
port=5060
nat=yes
canreinvite=no
qualify=no
disallow=all
allow=g729
dtmfmode=rfc2833
insecure=very
context=incoming
___
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Re: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hanging up
I have a PSTN termination provider foo which will accept standard U.S. calls in the form 110 digit ph#. I have an outbound route named foo, with dial pattern 5|., with the only entry in trunk sequence being IAX2/foo. I have an X-lite local extension, on which I can dial 5110 digit ph#, and asterisk will call out over foo and the phone at 10 digit ph# will ring. This rules out a lot of possible problems. extensions.conf includes this: [outgoingtest] exten = s,1,Playback(custom/testmsg) exten = s,2,Wait(1) exten = s,3,Hangup And yes, asterisk has been restarted since the last time any config files were modified. I have a test message at /var/lib/asterisk/sounds/custom/testmsg.gsm If I make the file 1.call containing: Channel: IAX2/foo MaxRetries: 1 RetryTime: 5 WaitTime: 10 Context: outgoingtest Extension: 110 digit ph# Priority: 1 and copy it to /var/spool/asterisk/outgoing/ then the phone doesn't ring, but this shows up on the asterisk console: -- Attempting call on IAX2/foo for 110 digit ph#@outgoingtest:1 (Retry 1) -- Hungup 'IAX2/foo-7' -- Attempting call on IAX2/foo for 110 digit ph#@outgoingtest:1 (Retry 2) -- Hungup 'IAX2/foo-8' The foo-7 and foo-8 on the console are different (numbers anywhere from 1 to 9) every time I try copying the file to outgoing. I tried using extension 5110 digit ph# instead of 110 digit ph# in 1.call, but that didn't work either. Why is it failing? Here's an update. With iax2 debugging enabled, when I copy 1.call to /var/spool/asterisk/outgoing/ here's what I get on the console: -- Attempting call on IAX2/foo for 110 digit ph#@outgoingtest:1 (Retry 1) Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 1 DCall: 0 [PSTN provider's IP:4569] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ulaw|alaw|gsm) CALLING PRESNTN : 67 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 LANGUAGE: en USERNAME: my username FORMAT : 64 CAPABILITY : 2097151 ADSICPE : 0 DATE TIME : 2006-05-02 13:39:26 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00016ms SCall: 00330 DCall: 1 [PSTN provider's IP:4569] AUTHMETHODS : 3 CHALLENGE : code USERNAME: my username Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00091ms SCall: 1 DCall: 00330 [PSTN provider's IP:4569] MD5 RESULT : c8cd34a533731f2ad50121395e0fe2a1 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00105ms SCall: 00330 DCall: 1 [PSTN provider's IP:4569] CAUSE : No such context/extension CAUSE CODE : 3 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00105ms SCall: 1 DCall: 00330 [PSTN provider's IP:4569] -- Hungup 'IAX2/foo-1' So far as I can tell, I'm properly following the directions listed at http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out So, based on the above, can anybody identify what's going wrong? Asterisk is saying No such context/extension, but clearly there is such a context, as shown in my files above. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PAP2/Sipura XML Provisioning File
Hi All, I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd PAP2-NA units all hooked up to Asterisk. As you can imagine, setting them up took a while, and changing settings on them also takes a while. In order to prepare for future deployments, I'd like to use XML provisioning (or any kind of remote provisioning). I figured since Sipura/Cisco won't release the utility to create the file unless you're a bigtime reseller, my only option is to use a XML file. Does anyone have Sipura/Linksys ATAs sample XML files? Thanks in advance for any help. Regards, Gonzalo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Extreme delay before * processes call files
Remco Barende wrote: Found it! It seems that Asterisk is looking at the date / time stamp of the call file to process the call?? I was simply moving the call files hoping it would just work (tm) I guess that the call files created on the samba share I created carried the time/date stamp of the local machine (workstation) and not the asterisk server causing a time difference. Now I run a touch * on the asterisk server before moving the call files, all the calls are now processed immediately. Is this intended behaviour for the call files?? Or just a bug? It's not a bug. Why not use NTP to sync the system's time to a well known source? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
