[Asterisk-Users] Sending DTMF after a call is set up
I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. I also want the caller to not hear any audio until the DTMF has been sent. This gets the caller to the right place and he doesnt have to hear the welcome message from the PBX. Dial apparently isnt the application to use. Is there something else or do I have to write my own? If the latter, can someone tell me what principal routines to call in this application? Thanks, Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending DTMF after a call is set up
The thing is I need to wait a few seconds. In fact, it's even worse, I need to wait a few seconds, dial an extension, wait a few more seconds, and then dial another! It's perfect for something like ... Wait(5) ... SendDTMF(123) ... Wait(5) ... SendDTMF(456) but the Dial command doesn't return until the call is done. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Webb Sent: Friday, February 18, 2005 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sending DTMF after a call is set up n Fri, 18 Feb 2005 16:06:54 -0500 Bill Hamlin [EMAIL PROTECTED] wrote: I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. I also want the caller to not hear any audio until the DTMF has been sent. This gets the caller to the right place and he doesnt have to hear the welcome message from the PBX. Dial apparently isnt the application to use. Is there something else or do I have to write my own? If the latter, can someone tell me what principal routines to call in this application? Thanks, Bill Not sure if it is what you are looking for, but look at this option in the wiki: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. You can find the entire list here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: n Fri, 18 Feb 2005 16:06:54 -0500 Bill Hamlin [EMAIL PROTECTED] wrote: I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. I also want the caller to not hear any audio until the DTMF has been sent. This gets the caller to the right place and he doesnt have to hear the welcome message from the PBX. Dial apparently isnt the application to use. Is there something else or do I have to write my own? If the latter, can someone tell me what principal routines to call in this application? Thanks, Bill [...] Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sending DTMF after a call is set up
Ah! You guys are right, the D option will do the trick I think. Thanks, Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson Sent: Friday, February 18, 2005 4:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sending DTMF after a call is set up Bill Hamlin wrote: I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. Try this posting: http://www.voip-info.org/wiki-Asterisk+cmd+dial?page=Asterisk%20cmd%20dialc omments_threshold=0comments_offset=0comments_sort_mode=commentDate_descco mments_maxComments=10comments_parentId=931#threadId1168 It might be channel specific, I do not know. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] restricting SIP access to asterisk
How do you set up Asterisk to allow SIP call requests from specific IP addresses? We have no control over what account (From: header) is used. We want to be able to allow calls based on the IP address the INVITE comes from, not the account. Is there a way to do that? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP client cannot connect to Asterisk
Try setting the SIP signalling port in your client to something other than 5060 (eg ) and run tethereal on your Asterisk box to see if you're getting packets on . -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of K Wong Sent: Monday, December 27, 2004 9:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP client cannot connect to Asterisk Hi: We have got SIP clients connecting to our Asterisk fine with a DSL connection behind router (NAT), but when we bring the Sipura 2000 ATA to a Rogers Cable connection behind a Netgear router (NAT), the SIP clients aren't able to reach the Asterisk at all. We enabled the SIP debug in Asterisk, and it doesn't see any request coming from these SIP clients, and we also tried the to use a XTEN Lite to connect to Simpletelecom within this network and it fails to register as well. It seems to be a network configuration problem, but there isn't much log in the router that we can dignose as the Netgear router WR814 only logs TCP web requests. Does anyone know what the problem could be? Does Rogers Cable blocks SIP ports? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astcc newbie question
I'm trying out ASTCC. I set the card length to 10, and generated a test card. 10 digits. I set the extensions file to: exten = 9175954700,1,Answer exten = 9175954700,2,DeadAGI(astcc.agi) exten = 9175954700,3,Hangup I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits. How come it thinks it is 12 digits? I set both the Published number and DID in the Brand to 9175954700. Was that the right thing to do? Maybe it's not recognizing the DID? Thanks, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as PSTN gateway
I've been asked to recommend a solution for a one-E1-port PSTN gateway supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know they work. I use the Asterisk software in a couple of places and would like to use the E100P. My question is whether anyone out there has any installations using this and what their opinion is about it (does it work? how's the audio quality? and so on). Thanks, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-way calling
I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help, Bill Hamlin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] question about CPU usage
I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Thanks, Bill Hamlin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] question about CPU usage
What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 200
You must have port mapping in the Linux NAT that allows the SIP-level packets to get to the * Server, so you need to add a port mapping for the RTP packets. I may be wrong but I think * sends RTP on the same port that it receives RTP on, so once the phone sends some RTP to * then the RTP coming back should work. Turn on sip debug to see the packets and cut and paste here if you're still having a problem. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Geert Nijpels Sent: Monday, March 22, 2004 4:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Snom 200 Barry Fawthrop wrote: Progress It seems I can't hear the Say Time, due to RTP Double NAT I'm guess this is both problem 1 and 2 really issue. My config: IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server ANyone know of work arounds the double NAT? or other methods to route RTP with snom 200s, to work around this? I think you can make progress with the following link: http://www.voip-info.org/tiki-index.php?page=NAT%20and%20VOIP Have fun, Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] question about CPU usage
Nope same problem. I just started it and did a couple of ps aux's and got this output: [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.6 1.3 115880 6676 ? R15:43 1:10 asterisk -vgcd root 20221 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.3 1.3 115880 6676 ? R15:43 1:13 asterisk -vgcd root 20223 0.0 0.1 3568 624 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.7 1.3 115880 6676 ? R15:43 1:16 asterisk -vgcd root 20225 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.4 1.3 115880 6676 ? R15:43 1:18 asterisk -vgcd root 20227 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.6 1.3 115880 6676 ? R15:43 1:20 asterisk -vgcd root 20229 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Monday, March 22, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage I think Steve is referring to the following line: export LD_ASSUME_KERNEL=2.4.1 If you put this in your command line before starting asterisk, you will get around the RH9 problem of leaving zombies when AGI processes quit. Other than that, I don't think it influences CPU load. Note that the line is not necessary for Fedora Core 1 regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin Sent: Monday, March 22, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem playing the first voice mail prompt
I dial an extension that starts up VoiceMailMain. When the call comes in the following lines are written to /var/log/messages: Feb 20 11:01:37 redhat2 kernel: Zapata Telephony Interface Registered on major 196 Feb 20 11:01:37 redhat2 kernel: No ISA tormenta card found at d Feb 20 11:01:37 redhat2 kernel: Zapata Telephony Interface Unloaded Feb 20 11:01:37 redhat2 insmod: /lib/modules/2.4.20-8/misc/torisa.o: init_module: Input/output error Feb 20 11:01:37 redhat2 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 Feb 20 11:01:37 redhat2 insmod: /lib/modules/2.4.20-8/misc/torisa.o: insmod char-major-196 failed With sip debug on the CLI interface I get the warning at line 521 in file.c, something about the file can't be written. I have a system that has no Zapata cards at all. Do I need to have one? Any ideas as to what may be wrong? Thanks, Bill Hamlin Globalnet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
