[asterisk-users] online active call watching
Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Regards - Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Project: AskoziaPBX
Greetings everyone, I've been working on a (yet another) all-in-one Asterisk based project. It is aimed at embedded / low power systems (but scales fine on more capable hardware) and is based on Asterisk 1.4.x and FreeBSD 6.2. Because of this, I've mostly been hanging out on the asterisk-bsd list as bugs rolled in and the system's features were improved. We're currently at public beta 10 after releasing pb1 in June and, I hope, ready to announce this to a bit larger audience. This is not a live-cd but rather an image that must initially be written to a disk, so a dedicated machine is needed. After that, the entire system is upgradeable through the webGUI. Anyone familiar with the m0n0wall project (http://m0n0.ch/wall) will feel right at home as AskoziaPBX was forked from it. Here are the quick facts from the website and a link to the page: * ~11 MB firmware image * PHP based GUI accessible via http(s) * based on Asterisk 1.4 and FreeBSD 6.2 * designed for embedded / low resource systems * images available for the following platforms: * generic pc * pc engines wrap * soekris net48xx * VMware player * GUI currently configures: * SIP, IAX, ISDN and Analog phones and providers * Conferencing * Voicemail (forwarded as e-mail attachment) * Call Groups * Call Parking * ...as well as all system settings (ntp, GUI port, etc.) * all configuration stored in a single XML file * Multilingual audio-prompts: * Dutch, English, French, German, Italian, Japanese, Russian, Spanish, Swedish * Multilingual voicemail notification e-mails: * Dutch, English, French, German, Italian, Polish, Spanish, Swedish site: http://askozia.com/pbx Thanks goes out to everyone in asterisk-bsd and pbx-users for testing / reporting and quite a few people in IRC who helped troubleshoot bugs as they popped up! (Also, please remember that this is still a beta.) Regards, -Michael I. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
On 9/10/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Hi with the CDR+mysql you can make query Invite+ack ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Behaviour
Thank you I will try tonight On 9/10/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita: On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. First of all, thank you for you reply The ATA is the Fritz!Box and I tried with different FW version but I have the same behaviour I have been using FritzBoxes for quite a while, and have not found such strange bugs - except after a Firmware Upgrade. It seems after some upgrades you need to do a factory reset (via the web interface) and enter your data again, else they behave stupidly. this is part of the sip.conf [180] type=peer username=180 secret=aa callerid=First180 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm [181] type=peer username=181 secret=bb callerid=Second181 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm Looks pretty OK to me. Just a stupid idea: Do you have a [general] section before those two? And then, I use type=friend, not type=peer, that _might_ make a difference in how asterisk matches sip.conf contexts to registered clients. 8 From my sip.conf: [sip501] mailbox=01 callerid=501 type=friend username=sip501 secret=lk1j2eu89 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw [sip502] mailbox=02 callerid=502 type=friend username=sip502 secret=1092jd0 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw =8 Note: Those two accounts belong to the same FritzBox. I tried to switch the account for the two ports but what it is important is only the order in the sip.conf That made me think about that friend/peer thingy. I found some information in german and I do not know it The FritzBoxes are popular here in Germany - no wonder, being a German manufactured product and being given away for (nearly) free with any 2-year DSL contract... I like them nevertheless :) BR, HTH Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
On Sun, Sep 09, 2007 at 11:37:03PM -0700, satish patel wrote: Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to You can list the channels of Asterisk. While channels are not exactly calls (a call can span over two channels), it gives you a good idea. An occasional 'show channels' from the CLI, a terminal with: watch asterisk -n -rx 'show channels' and the astman tool included with Asterisk are basically that. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USA Termination
Send us your traffic, we can terminate it in the USA for you --- $.00475 US TERMINATION. International Origination Traffic sent with international CLI* 1/1 Billing 50,000/day $.006/minute 100,000/day $.00575/minute 250,000/day $.00555/minute 500,000/day $.0050/minute 1,000,000/day $.00475/minute off-net traffic$.011/minute statsASR 87% ACD 9+ G711/729SIP or H323. Therefore on-net % will increase. Unlimited Port Capacity. Our footprint is largest. Unlimited Port Capacity. Our footprint is largest. Send us your CDRs we will analyze for on-net and off-nettraffic ratio. US CLI can replace international CLI sent, Unlimited capacity, SIP or H323, G711 or G729. Email me off list - [EMAIL PROTECTED] Claude +1 954 905 8612 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
satish patel wrote: Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Flash Operator Panel is what you'd want to look at: http://www.asternic.org Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? G.711ulaw and G.711alaw are the audio transmission methods used for ISDN. If you have a T1 line then the transmission method is G.711ulaw. I've been told that if you play a ulaw signal down an alaw line (T1 signal down E1) then at the other end the voice sounds a bit like a dalek. (Iit's very hard to do this with asterisk since it automatically transcodes between endpoints). The lack of a performance hit is quite striking when you have a recording playing back as a native format rather than being transcoded. (well, it's quite striking when you have thousands of them running simultaneously). Bart Steve Totaro wrote: Michiel van Baak wrote: On 10:28, Sun 09 Sep 07, Barton Fisher wrote: I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram depends on what codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. The answer is going to ulaw or alaw depending where you live. T1 should most likely be using ulaw so make everything ulaw, end to end. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2516 (20070909) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 56k modem configuration
Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cascading queues calls not joining unavailable queues.
Quoting Mark Michelson ([EMAIL PROTECTED]): -- Called SCCP/231 -- Called SCCP/220 -- SCCP/220-009b is busy -- SCCP/231-009a is busy I'd like asterisk to quit trying when all agents are busy, but i don't think it's possible without scripting it yourself with some AGI-script that checks 'show queues' output. It sounds as though skinny devices may not be reporting their device state correctly, and so the queue believes that the devices are available. Looking at the output of 'show queues' everything looks completely OK when i put the phone in various states of 'being available'. I think it's more an opinion on what 'unavailable' is. Or perhaps they are reporting a state that the queue does not know about. If this is the case, we may be dealing with a bug. I will test locally when I can get access to a Skinny phone and see what's going on. We're using chan_sccp.so in combination with Cisco 796x phones (With CTU ringtone! Whee! :P). Maybe it doesn't really work right because of this, but as Asterisk *tells me* it knows nobody is answering a queue, i wonder why it keeps trying ;-) Kind regards, Sander. -- | If you jog backwards, will you gain weight? | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cascading queues calls not joining unavailable queues.
Quoting James FitzGibbon ([EMAIL PROTECTED]): Unfortunately, the patches weren't done against trunk or the head of 1.4, and the author didn't file a disclaimer with Mantis, so the bug ( http://bugs.digium.com/view.php?id=9165) was recently closed. That's just too bad, as this might be a solution to our 'problems'. :) -- | The less hair I have, the more head I get! | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the difference between increasing theverbose level and the debug level?
I just want to add that it is the best way to learn. Till today I thank those on the list that told me to stay away from GUI's and learn the real asterisk. If you still can't figure out the difference I can help you out but it is better if you learn on your own. - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 10, 2007 2:42 AM Subject: Re: [asterisk-users] What is the difference between increasing theverbose level and the debug level? In general keep in mind, asterisk is very user friendly and wont bite :). Trial and error is a good friend to get to know asterisk so that you know what all of these mean. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; What is the difference between increasing the verbose level and the debug level? By increasing the verbose level, then I will get more traces messages and by increasing the debug level, I will also get more traces messages. So what is the difference? Any help? Regards Bilal Ghayad Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA Termination
There is a Biz list for a reason. Please look at the emails headers Non-Commercial Discussion - Original Message - From: Claude Cunningham [EMAIL PROTECTED] To: Commercial and Business-Oriented Asterisk Discussion [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 10, 2007 12:09 PM Subject: [asterisk-users] USA Termination Send us your traffic, we can terminate it in the USA for you --- $.00475 US TERMINATION. International Origination Traffic sent with international CLI* 1/1 Billing 50,000/day $.006/minute 100,000/day $.00575/minute 250,000/day $.00555/minute 500,000/day $.0050/minute 1,000,000/day $.00475/minute off-net traffic$.011/minute statsASR 87% ACD 9+ G711/729SIP or H323. Therefore on-net % will increase. Unlimited Port Capacity. Our footprint is largest. Unlimited Port Capacity. Our footprint is largest. Send us your CDRs we will analyze for on-net and off-nettraffic ratio. US CLI can replace international CLI sent, Unlimited capacity, SIP or H323, G711 or G729. Email me off list - [EMAIL PROTECTED] Claude +1 954 905 8612 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: Thanks. My results after applying the patch and recompiling are that the problem can only be replicated with calls from mobile networks. Digits like 160 entered in the digital receptionist by a caller are received by the asterisk server as 16660 sometimes. Other times it is received as 1660. Digits like 1234 are received as 1222334 etc... From fixed lines, there is no problem. Digits are received as they have been sent. Any other pointers? Hmm, that sounds like a problem with the GSM-to-PSTN gateway that the calls are passing through. Unless things are different in Uganda, I believe when a user presses a DTMF key on their mobile, it doesn't send a tone through the mobile network, but rather a start dtmf control message followed by a stop dtmf control message. When the call gets gatewayed from GSM to the PSTN network, it is the job of the gateway to generate the tones as instructed by the control protocol. (Someone please correct me if I'm wrong). So you may need to take it up with your telco. Cheers Tony Thanks a lot. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, September 09, 2007 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote: I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. A make install will not overwrite any configfile. It will install the asterisk binary and the modules (thus overwriting the existing files) but configfiles will only be overwritten when you run: make samples -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the difference between increasingtheverbose level and the debug level?
Except in the cases where what you observe in real life is buggy behaviour, and not what the designer/implementor intended. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: 10 September 2007 12:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the difference between increasingtheverbose level and the debug level? I just want to add that it is the best way to learn. Till today I thank those on the list that told me to stay away from GUI's and learn the real asterisk. If you still can't figure out the difference I can help you out but it is better if you learn on your own. - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 10, 2007 2:42 AM Subject: Re: [asterisk-users] What is the difference between increasing theverbose level and the debug level? In general keep in mind, asterisk is very user friendly and wont bite :). Trial and error is a good friend to get to know asterisk so that you know what all of these mean. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; What is the difference between increasing the verbose level and the debug level? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Also your Disk subsystem speed. having disk RAM , makes sense in your case. On 9/10/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? G.711ulaw and G.711alaw are the audio transmission methods used for ISDN. If you have a T1 line then the transmission method is G.711ulaw. I've been told that if you play a ulaw signal down an alaw line (T1 signal down E1) then at the other end the voice sounds a bit like a dalek. (Iit's very hard to do this with asterisk since it automatically transcodes between endpoints). The lack of a performance hit is quite striking when you have a recording playing back as a native format rather than being transcoded. (well, it's quite striking when you have thousands of them running simultaneously). Bart Steve Totaro wrote: Michiel van Baak wrote: On 10:28, Sun 09 Sep 07, Barton Fisher wrote: I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram depends on what codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. The answer is going to ulaw or alaw depending where you live. T1 should most likely be using ulaw so make everything ulaw, end to end. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2516 (20070909) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF bug in dsp.c and 1.4.11
In article [EMAIL PROTECTED], Jerry Geis [EMAIL PROTECTED] wrote: I was wondering if this bug: http://bugs.digium.com/view.php?id=10535 would affect a PRI connection. I seem to be dropping DTMF digits on the PRI. The company says they have test the line and they way the PRI is fine as far as they are concerned. So will this bug and patch help me? I am running 1.4.11 Yes, that bug was submitted by me, and it was a PRI on which I was having the problems. If there is a slight bounce on the leading edge of a digit, then it can easily be dropped altogether by Asterisk. The patch fixes that, and also adds debouncing of the trailing edge (else a trailing bounce might give a double-digit). Give it a try - I expect it will help a lot. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
Which Panasonic PBX? On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I am having Asterisk pbx which is running without any problem now i want to connect this with Panasonic pbx with FXS port so, if any body want to call panasonic users than he will call or vise-versa. i want to connect only two extension with Asterisk so, all communication done only on these two line. what is the process and what is the setting in sip.conf and extensions.conf to communicate with Asterisk and Panasonic pbx. Rajeev. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
Tom, The device is voxbone from voxbone.com . I am using a DID as an access number...it worked with same config with asterisk 1.2.12 and a2billing 1.2.3, but doesn't work with asterisk 1.4.11 and a2billing 1.3 Can you tell me what am I missing? Apa Tom Lynn [EMAIL PROTECTED] wrote: I suspect if you remove the callerid entry from this device's sip.conf definition things will work better. On 9/9/07, Apa Minerala [EMAIL PROTECTED] wrote: I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call 778f89593967725f0abe40eb1752504c no reply to our critical packet. What is the critical packet that is not being responded to? Please help. - Pinpoint customers who are looking for what you sell. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
So I'll rephrase to some devices will not operate properly, since after your message I am assuming that you tested this with most devices. On 9/10/07, Benjamin Jacob [EMAIL PROTECTED] wrote: C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
try the astman command. __Yehavi: ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
On 9/9/07, Barton Fisher [EMAIL PROTECTED] wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? If you absolutely want to be sure, use 'pri intense debug span X' and watch for SETUP messages: Protocol Discriminator: Q.931 (8) len=62 Call Ref: len= 2 (reference 542/0x21E) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 95] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 21 ] You'll see the voice characteristics in the Bearer Capability details (I have NI-2, this might be different for NI-2 or other PRI variants). But as others have mentioned, generally T1 PRI = uLaw, E1 PRI = aLaw. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
Actually this problem is with a telco in the US [the setup is in the US]. I will get in touch with them to have them look into it. There is another similar setup with the same telco and there are no such problems. The only difference in the setups is that in this case, the T1 is terminated into a Cisco 2430 Integrated Access Device and then a T1 from that device terminates into the Asterisk PBX. Probably I will have them bypass the Cisco device and see whether I can replicate this again. Joseph. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, September 10, 2007 7:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: Thanks. My results after applying the patch and recompiling are that the problem can only be replicated with calls from mobile networks. Digits like 160 entered in the digital receptionist by a caller are received by the asterisk server as 16660 sometimes. Other times it is received as 1660. Digits like 1234 are received as 1222334 etc... From fixed lines, there is no problem. Digits are received as they have been sent. Any other pointers? Hmm, that sounds like a problem with the GSM-to-PSTN gateway that the calls are passing through. Unless things are different in Uganda, I believe when a user presses a DTMF key on their mobile, it doesn't send a tone through the mobile network, but rather a start dtmf control message followed by a stop dtmf control message. When the call gets gatewayed from GSM to the PSTN network, it is the job of the gateway to generate the tones as instructed by the control protocol. (Someone please correct me if I'm wrong). So you may need to take it up with your telco. Cheers Tony Thanks a lot. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, September 09, 2007 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote: I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. A make install will not overwrite any configfile. It will install the asterisk binary and the modules (thus overwriting the existing files) but configfiles will only be overwritten when you run: make samples -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broken UDP streams
Maximum retries exceeded on transmission usually comes from NAT issues. you can try this system without NAT and see if problem has resolved. On 9/7/07, Adrian Marsh [EMAIL PROTECTED] wrote: Hi All, I'm working from home today (DSL - Internet - 2MB leased line - A*K server behind NAT), and trying to pickup voicemail using Zoiper.. I can access the VM system, I hear all the prompts, and I can even hear part of the message playback. But then I get silence on the call (call stays up), and I get: Parsing '/var/spool/asterisk/voicemail/default/2027/Old/msg.txt': Found -- Playing '/var/spool/asterisk/voicemail/default/2027/Old/msg' (language 'en') Sep 7 13:51:30 WARNING[30737]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission NmM3YmNhNjk0NzhhMjFlYmU5Yzg1YTBmNThlZDNhYWQ. for seqno 2 (Critical Response) Sep 7 13:51:30 WARNING[30737]: chan_sip.c:1245 retrans_pkt: Hanging up call NmM3YmNhNjk0NzhhMjFlYmU5Yzg1YTBmNThlZDNhYWQ. - no reply to our critical packet. == Spawn extension (from-sip, voicemail, 4) exited non-zero on 'SIP/427-b780fa40' On the A8k log. I'm guessing packets are getting lost, but don't understand why it would only be in VM playback that it happens. Any ideas? Adrian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
C F wrote: BTW, AFAIK, there is no such thing as host=static it's either dynamic or an IP/Name. Yeah, I learned that the hard way. I had only set up dynamic devices for a couple of months, and the first time I had reason to set up a device with a static IP, I just assumed that 'host=static' would work in sip.conf. Dur, it took me a couple of hours to figure out why my fax machine could fax, but not receive. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Thanks Guys... ulaw it is. One more question if you don't mind. If a phase recorded as both .wav and .ulaw in the same folder, which will asterisk pick using Playback(), Read() and Background() since you can't specify the file extension in the command? I thought I change my script to begin recording new messages in ulaw instead of converting them all to ulaw at once. So it's possible to have two prompts with both file extension at a time Bart Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1 (rest of world) 99.9% of the time. Unless you have something strange or different, I'd record in ulaw for T1. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx aFwrtGNKZ0EbZr176MDZUkY= =HvDo -END PGP SIGNATURE- __ NOD32 2517 (20070910) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failover SIP logic
I need some extensions logic assistance, I'm trying to dial out one of multiple
SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call
per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the 1st
trunk is busy)
extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
[from-internal]
exten = _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
exten = _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
sip.conf:
[trunk1]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk2]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk3]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
Here's asterisk output when someone dials out:
Executing [EMAIL PROTECTED]:1] Macro(SIP/6001-007e2840,
trunkdial|+1xx) in new stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6001-007e2840,
SIP/trunk1/+1xx) in new stack
[Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to
peer 'trunk1' rejected due to usage limit of 1
-- Couldn't call trunk1/+1xx
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/6001-007e2840, ) in new
stack
I don't want the dialplan to cascade like:
exten = 1,dial...
exten = 2,dial...
Because if the remote end hangs up I don't want it going to priority 2 to dial
out again(in case my user doesn't hit hang-up on their end) so I need logic to
detect a busy channel and jump to the next section..
Thanks for any help.
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Re: [asterisk-users] Register Extension
On 7 Sep 2007, at 17:56, phananhvu wrote: I means i want to use a software library to write a program that register an extension to Asterisk system. After that, i can bind my IP Phone to that extension. I wonder if Asterisk-Java can deal with this ?? Ah, you mean create an extension that a phone can register with ? Last time I looked, the answer is no, Asterisk-java doesn't help you create entries in extensions.conf or sip.conf . The way I've done this in java is to map sip.conf (In my case iax.conf) to a database table (see extconfig.conf). Then have your java write to that database table using JDBC. After some trouble I even got it working with Oracle. Tim. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
It will automatically pick the best recording for the current codec, so if you are in ulaw, it will choose the ulaw prompt. Barton Fisher wrote: Thanks Guys... ulaw it is. One more question if you don't mind. If a phase recorded as both .wav and .ulaw in the same folder, which will asterisk pick using Playback(), Read() and Background() since you can't specify the file extension in the command? I thought I change my script to begin recording new messages in ulaw instead of converting them all to ulaw at once. So it's possible to have two prompts with both file extension at a time Bart -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover SIP logic
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Jeremy Mann wrote:
I need some extensions logic assistance, I'm trying to dial out one of
multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
allow 1 call per trunk) and roll over to a second or third depending on that
busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the
1st trunk is busy)
extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
[from-internal]
exten = _NXXNXX,1,Macro(trunkdial,+1${EXTEN})
exten = _1NXXNXX,1,Macro(trunkdial,+${EXTEN})
sip.conf:
[trunk1]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk2]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
[trunk3]
host=xxx.xxx.xxx.xxx
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
reinvite=no
nat=no
fromuser=+xxx
call-limit=1
Here's asterisk output when someone dials out:
Executing [EMAIL PROTECTED]:1] Macro(SIP/6001-007e2840,
trunkdial|+1xx) in new stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6001-007e2840,
SIP/trunk1/+1xx) in new stack
[Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to
peer 'trunk1' rejected due to usage limit of 1
-- Couldn't call trunk1/+1xx
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/6001-007e2840, ) in new
stack
I don't want the dialplan to cascade like:
exten = 1,dial...
exten = 2,dial...
Because if the remote end hangs up I don't want it going to priority 2 to
dial out again(in case my user doesn't hit hang-up on their end) so I need
logic to detect a busy channel and jump to the next section..
If you have this:
exten = _X.,1,Dial(SIP/trunk1)
exten = _X.,2,Dial(SIP/trunk2)
exten = _X.,3,Dial(SIP/trunk3)
then, only if trunk is busy, will it go to trunk2, if thats busy, it will go to
trunk 3.
Reason is, is that control wont return to the dial plan(except h) if the call
was
successfull. SO if the call went through on trunk 1, then it will exit, not
dial trunk2
or trunk3. So this dial plan will work. But its very sequential, i.e. will
try trunk1,
then trunk2, then trunk3. If you want to replicate round-robin, r, then do
this:
[globals]
IPt=trunk1-trunk2-trunk3
COUNTt=0
NoOfChannels=3
[just-an-idea]
exten = _X.,1,Gotoif($[${COUNTt} = ${NoOfChannels}] ? 2:3)
exten = _X.,2,SetGlobalVar(COUNTt=0])
exten = _X.,3,SetGlobalVar(COUNTt=$[${COUNTt}+1])
exten = _X.,4,Set(tr=${CUT(IPt,-,${COUNTt})})
exten = _X.,5,Dial(SIP/tr/${EXTEN})
modify at your leisure. So if you get a few more trunks, you just change
NoOfChannels
--
thanks,
Yusuf
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Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
On 9/10/07, Barton Fisher [EMAIL PROTECTED] wrote: Thanks Guys... ulaw it is. One more question if you don't mind. If a phase recorded as both .wav and .ulaw in the same folder, which will asterisk pick using Playback(), Read() and Background() since you can't specify the file extension in the command? I thought I change my script to begin recording new messages in ulaw instead of converting them all to ulaw at once. So it's possible to have two prompts with both file extension at a time Asterisk will try to find file in codec currently in use, and if it can't find, it will try to use file with less translation time (try show transcoding in CLI). So - you can have files in all the codecs used in your PBX, asterisk will choose most appropriate. The same goes for MOH. A little caveat - sox doesn't understands file extensions used by asterisk (or it's just asterisk, trying to use file extensions that match codec name). So - some sox commandline hints: ulaw: -t ul alaw: -t al slin: -t raw -s -w Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
Hello, 2007/9/10, C F [EMAIL PROTECTED]: Which Panasonic PBX? On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I am having Asterisk pbx which is running without any problem now i want to connect this with Panasonic pbx with FXS port so, if any body want to call panasonic users than he will call or vise-versa. How ? Do you plan to dedicate Panasonic PBX FXS ports to act as a trunk or would dedicate one port for each Asterisk user ? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Atis wrote: A little caveat - sox doesn't understands file extensions used by asterisk (or it's just asterisk, trying to use file extensions that match codec name). So - some sox commandline hints: ulaw: -t ul alaw: -t al slin: -t raw -s -w Or (since 1.4.0) in the asterisk cli type: Convert /path/to/filename.wav /path/to/filename.ulaw ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
I need some extensions logic assistance, I'm trying to dial out one of
multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
allow 1 call per trunk) and roll over to a second or third depending on that
busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the
1st trunk is busy) extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.
HTH,
--
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945
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Re: [asterisk-users] Failover SIP logic
Asterisk 1.4.11
Sorry, meant to include that
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic
Ciao Jeremy,
I need some extensions logic assistance, I'm trying to dial out one of
multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only
allow 1 call per trunk) and roll over to a second or third depending on that
busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the
1st trunk is busy) extensions.conf:
[globals]
trunk_1 = SIP/trunk1
trunk_2 = SIP/trunk2
trunk_3 = SIP/trunk3
[macro-trunkdial]
exten = s,1,Dial(${trunk_1}/${ARG1})
exten = s,2,Hangup()
exten = s,102,Dial(${trunk_2}/${ARG1})
exten = s,103,Hangup()
exten = s,203,Dial(${trunk_3}/${ARG1})
exten = s,204,Hangup()
Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.
HTH,
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Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
On 08:04, Mon 10 Sep 07, Barton Fisher wrote: Thanks Guys... ulaw it is. One more question if you don't mind. If a phase recorded as both .wav and .ulaw in the same folder, which will asterisk pick using Playback(), Read() and Background() since you can't specify the file extension in the command? I thought I change my script to begin recording new messages in ulaw instead of converting them all to ulaw at once. So it's possible to have two prompts with both file extension at a time It will use the one for the channel codec. So you can have a file in every format and asterisk will pick the one that matches the channel codec. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemans SIP/PSTN phone S450
Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see Got SIP response 405 Method Not Allowed back from 192.168.3.64 but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. ubiphone*CLI -- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat keepalive --- ubiphone*CLI -- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat keepalive --- -- Got SIP response 489 Bad event back from 192.168.3.10 ubiphone*CLI -- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat keepalive --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.3.64:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport From: asterisk sip:[EMAIL PROTECTED];tag=as35c7a074 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 10 Sep 2007 17:23:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- ubiphone*CLI -- SIP read from 192.168.3.64:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as35c7a074 To: sip:[EMAIL PROTECTED]:5060;tag=1624959632 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Contact: Adrian Marsh sip:[EMAIL PROTECTED]:5060 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' ubiphone*CLI -- SIP read from 192.168.3.64:5060: REGISTER sip:some.server.com SIP/2.0 Via: SIP/2.0/UDP 192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;rport From: Adrian Marsh sip:[EMAIL PROTECTED];tag=3054246604 To: Adrian Marsh sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 291 REGISTER Contact: Adrian Marsh sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 User-Agent: S450 IP0207 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.3.64 : 5060 (NAT) Transmitting (NAT) to 192.168.3.64:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;receive d=192.168.3.64;rport=5060 From: Adrian Marsh sip:[EMAIL PROTECTED];tag=3054246604 To: Adrian Marsh sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 291 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 192.168.3.64:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;receive d=192.168.3.64;rport=5060 From: Adrian Marsh sip:[EMAIL PROTECTED];tag=3054246604 To: Adrian Marsh sip:[EMAIL PROTECTED];tag=as5908b79f Call-ID: [EMAIL PROTECTED] CSeq: 291 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3960830f Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms ubiphone*CLI -- SIP read from 192.168.3.64:5060: REGISTER sip:some.server.com SIP/2.0 Via: SIP/2.0/UDP 192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;rport From: Adrian Marsh sip:[EMAIL PROTECTED];tag=3054246604 To: Adrian Marsh sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 292 REGISTER Contact: Adrian Marsh sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=6627, realm=asterisk, algorithm=MD5, uri=sip:some.server.com, nonce=3960830f, response=7e032e9766f943e9f60f7d1f46114dee Max-Forwards: 70 User-Agent: S450 IP0207 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.3.64 : 5060 (NAT) Transmitting (NAT) to 192.168.3.64:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;received =192.168.3.64;rport=5060 From: Adrian Marsh sip:[EMAIL PROTECTED];tag=3054246604 To: Adrian Marsh sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 292 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 192.168.3.64:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;received =192.168.3.64;rport=5060 From: Adrian Marsh sip:[EMAIL PROTECTED];tag=3054246604 To: Adrian Marsh sip:[EMAIL PROTECTED];tag=as5908b79f Call-ID: [EMAIL PROTECTED] CSeq: 292 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 180 Contact:
[asterisk-users] Partitioning DSL input
Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
pfSense works very well for this. You can use it to setup VLANs (one for your PCs, the other for your VoIP equipment), and it has a traffic shaping/queuing mechanism for prioritizing VoIP. AR On 9/10/07, C. Savinovich [EMAIL PROTECTED] wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ubuntu Feisty
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christian wrote: Hello, On 2007-09-09 at 22:36 Ron Wellsted wrote: Christian wrote: Hi, What parameter should I use to that command? On 2007-09-09 at 13:45 Ron Wellsted wrote: Tzafrir Cohen wrote: On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote: Hi all, Have just installed v1.4.11 of Asterisk, but I am trying to have it start at boot but with no luck. I have used the make config command but it doesn't start. Any help would be apreciated, many thanks! use the command update-rc.d Also, as always in the case of software that has already been packaged, it may help to look at the existing package. I used update-rc.d asterisk 30 to ensure that it started after zaptel and mysql (which by default start at 20). Sorry, it should have read sudo update-rc.d asterisk defaults 30 Many thanks, will try that. Is Zaptel already loaded or will I need to do another command for that? Still learning. Many thanks, Christian Zaptel will need to be loaded if needed for hardware and/or timing. In the Zaptel source directory, there is zaptel.init, modify this for your /etc/init.d/zaptel file - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.136111 Linux Counter No. 202120 Ekiga: 645022 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRuWM3ktP/KMNOfRbAQJpSwf9HmC6lP/DI7/ych2CKKQz6Nk2/S8pJoy+ 2btD75+tyhImepe03KOeQyWuWu0HLhJW7pakrIFov3Ey7gwX4rqj+z9sr1r/goA4 JGkYJZN4cRFeZZpgN0YTXAbQpWSaXKwxjXJI6i3va3vEk9h1csXzFlKPZQHDtBI8 3ByJveuAmifBC6+5DBhO2nSFBAZhhPZjfN02ggLoZMW6hCIJH52iL+tw/2NVNY8e Wqw/Fe/FyAt6H10N4Zud6WtRdSQdJ0sF5ZMJ4qCgPG683oUgsYF+sdTSt00KMwAF OX73EKPyHmrBwIOmn8w/cupLN7Hx7MDhXC2dTpqGvuVVArAWcL9swA== =ZkRf -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
C. Savinovich wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich Check the recent archives. Someone announced that they had a Beta package for the WRT54G (and possibly other 3rd party compatible firmware routers) that would achieve exactly that. Beyond that, checkout 3rd party firmwares that run on these routers, some have QoS and traffic shaping abilities. Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
Looks good. a lot of initial work, but looks worth the effort. Do you find that it improves the quality of your VOIP calls? C. Savinovich From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar Sent: Monday, September 10, 2007 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] Partitioning DSL input pfSense works very well for this. You can use it to setup VLANs (one for your PCs, the other for your VoIP equipment), and it has a traffic shaping/queuing mechanism for prioritizing VoIP. AR On 9/10/07, C. Savinovich [EMAIL PROTECTED] wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). Many Thanks C. Savinovich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
Thanks for answering guys! Ok, let me see if i understood. If I use the line tapping strategy I wont be able to use asterisk to do the recordings. Correct? So, i need to use the asterisk as the Man in the Middle ( I think that's the same as the back to back suggestion from Tzafrir, Isn't it? ). Ok, so every call will pass through Asterisk and I can do anything i want with it. Thats cool, but since all the calls pass through my recording box I've just created another fail point. And if someday my recording box stop responding? Is there someway to minimize that? TIA, Ricardo On 9/5/07, Andrew Latham [EMAIL PROTECTED] wrote: or a man in the middle... http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle On 9/5/07, Steve Totaro [EMAIL PROTECTED] wrote: Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- haoole alea jacta est ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemans SIP/PSTN phone S450
On Mon, 10 Sep 2007, Adrian Marsh wrote: Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see Got SIP response 405 Method Not Allowed back from 192.168.3.64 but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. I have several Siemens C460IP's on various servers... They all do the same thing too. Doesn't seem to have any adverse effect though. Gordon ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
You could buy two identical servers and use the device (name escapes me) that will detect one server going down and flip the ISDN traffic to the spare. Or you could just buy a really good server with redundant power supplies, raid 5, and hope for the best. Thanks, Steve Ricardo Gemignani wrote: Thanks for answering guys! Ok, let me see if i understood. If I use the line tapping strategy I wont be able to use asterisk to do the recordings. Correct? So, i need to use the asterisk as the Man in the Middle ( I think that's the same as the back to back suggestion from Tzafrir, Isn't it? ). Ok, so every call will pass through Asterisk and I can do anything i want with it. Thats cool, but since all the calls pass through my recording box I've just created another fail point. And if someday my recording box stop responding? Is there someway to minimize that? TIA, Ricardo On 9/5/07, *Andrew Latham* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: or a man in the middle... http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle On 9/5/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] */ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- haoole alea jacta est ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco UC 500
Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
On 9/10/07, Ira [EMAIL PROTECTED] wrote: At 02:11 PM 9/10/2007, you wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). dd-wrt or Sveasoft on a Linksys router though I understand there are better choices in routers today. Don't expect too much out of traffic shaping. While it should work nearly perfectly upstream, there's only so much you can do to control the downstream (from your ISP to you). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
Thanks Steve, If somebody knows about this hardware, or already used it. Please give me some help. TIA, Ricardo On 9/10/07, Steve Totaro [EMAIL PROTECTED] wrote: You could buy two identical servers and use the device (name escapes me) that will detect one server going down and flip the ISDN traffic to the spare. Or you could just buy a really good server with redundant power supplies, raid 5, and hope for the best. Thanks, Steve Ricardo Gemignani wrote: Thanks for answering guys! Ok, let me see if i understood. If I use the line tapping strategy I wont be able to use asterisk to do the recordings. Correct? So, i need to use the asterisk as the Man in the Middle ( I think that's the same as the back to back suggestion from Tzafrir, Isn't it? ). Ok, so every call will pass through Asterisk and I can do anything i want with it. Thats cool, but since all the calls pass through my recording box I've just created another fail point. And if someday my recording box stop responding? Is there someway to minimize that? TIA, Ricardo On 9/5/07, *Andrew Latham* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: or a man in the middle... http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle On 9/5/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] */ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- haoole alea jacta est ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- haoole alea jacta est ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco UC 500
Jeremy Mann wrote: Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. Google cisco UC500, hit #2 = http://www.cisco.com/en/US/products/ps7293/products_data_sheet0900aecd8061fb06.html Quotes: Core components of the Cisco Unified Communications 500 Series include:Cisco Unified IP phones, including wireless handsets and Session Initiation Protocol (SIP) phones PSTN interfaces and features: SIP trunks and RFC 2833 support Does that help? I'll bet Asterisk is cheaper though. :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
http://www.voipsupply.com/manufacturers/RedFone_Communications.html?gclid=CKmd5OrbuY4CFVB1OAodfC7PxQ Ricardo Gemignani wrote: Thanks Steve, If somebody knows about this hardware, or already used it. Please give me some help. TIA, Ricardo On 9/10/07, *Steve Totaro* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You could buy two identical servers and use the device (name escapes me) that will detect one server going down and flip the ISDN traffic to the spare. Or you could just buy a really good server with redundant power supplies, raid 5, and hope for the best. Thanks, Steve Ricardo Gemignani wrote: Thanks for answering guys! Ok, let me see if i understood. If I use the line tapping strategy I wont be able to use asterisk to do the recordings. Correct? So, i need to use the asterisk as the Man in the Middle ( I think that's the same as the back to back suggestion from Tzafrir, Isn't it? ). Ok, so every call will pass through Asterisk and I can do anything i want with it. Thats cool, but since all the calls pass through my recording box I've just created another fail point. And if someday my recording box stop responding? Is there someway to minimize that? TIA, Ricardo On 9/5/07, *Andrew Latham* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: or a man in the middle... http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle On 9/5/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager API - Originate command
Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Line Tapping
I think they mean the Rhino Dax... http://rhinoequipment.com/minidax.html On 9/10/07, Steve Totaro [EMAIL PROTECTED] wrote: http://www.voipsupply.com/manufacturers/RedFone_Communications.html?gclid=CKmd5OrbuY4CFVB1OAodfC7PxQ Ricardo Gemignani wrote: Thanks Steve, If somebody knows about this hardware, or already used it. Please give me some help. TIA, Ricardo On 9/10/07, *Steve Totaro* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You could buy two identical servers and use the device (name escapes me) that will detect one server going down and flip the ISDN traffic to the spare. Or you could just buy a really good server with redundant power supplies, raid 5, and hope for the best. Thanks, Steve Ricardo Gemignani wrote: Thanks for answering guys! Ok, let me see if i understood. If I use the line tapping strategy I wont be able to use asterisk to do the recordings. Correct? So, i need to use the asterisk as the Man in the Middle ( I think that's the same as the back to back suggestion from Tzafrir, Isn't it? ). Ok, so every call will pass through Asterisk and I can do anything i want with it. Thats cool, but since all the calls pass through my recording box I've just created another fail point. And if someday my recording box stop responding? Is there someway to minimize that? TIA, Ricardo On 9/5/07, *Andrew Latham* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: or a man in the middle... http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle On 9/5/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ricardo Gemignani wrote: Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani Check out OrecX but you should be able to record that volume of calls natively on the box (that is assuming you are using Asterisk as your call center system. Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
At 02:11 PM 9/10/2007, you wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). dd-wrt or Sveasoft on a Linksys router though I understand there are better choices in routers today. Ira ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API - Originate command
On 9/11/07, Wai Wu [EMAIL PROTECTED] wrote: Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. What did you mean by simultaneously? Opening 120 manager connections, and originating call at exactly the same time? I doubt.. So, probably there is some interval - within second/minute, etc.. And how many manager connections do you use? Maybe asterisk have some limit of them. Also - i think, there is some limit of asterisk accepting commands sequentially from one connection. Btw, what is your CPU load, when creating those 120 calls instantly? Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API - Originate command
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Wai Wu wrote: Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. First off, you should be using Async: true Secondly, you shouldn't really be doing 120 simultaneous calls. If your server can take say 300 concurrent calls, you will probably need to start those up with about 30ms between them. If you really need to start 120 calls all at the identical time, you probably want to be looking at clustering Asterisk servers. In SmoothTorque we set a minimum value for delay between calls, then have a funnel which accepts calls from predictive campaigns. The funnel knows about the connections to the Asterisk servers, and distributes the calls in a round robin fashion (assuming all servers are up). Each server has a queue which allows calls sent to that server to back up, and if a queue gets too full calls won't be sent to that server. If, after stopping the sending of calls to a server, the queue does not empty out, the server is placed into an inactive state, and a server marked as standby is moved into the active state. Any calls which remain in the queue are redistributed to other servers. Hope that helps! - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c T3+G284pc4LV/JMlj13v8gU= =oaJj -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Thanks, again. That did the trick! Bart Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? Basically its going to be g711.ulaw for T1 (USA) and g711.alaw for E1 (rest of world) 99.9% of the time. Unless you have something strange or different, I'd record in ulaw for T1. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5MI9DQNt8rg0Kp4RAgRWAKCL2l8egvLV2Xu3T754KJMzGXrKnQCfboCx aFwrtGNKZ0EbZr176MDZUkY= =HvDo -END PGP SIGNATURE- __ NOD32 2517 (20070910) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF
Hi Ever since I upgraded to the most recent V1.2 * and Zaptel DTMF stopped working. If I call my cell and press a key, I can hear that it's trying to send a tone, but there's not enough to trigger the menus at the places I call. I can't see that this is user adjustable and it use to work just fine. Any suggestions on how to fix or troubleshoot this. I did recently install * and Zaptel 1.4 and then go back to 1.2 if that matters. Thanks ever so much, Ira ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA - How to detect software failure?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yann JOUANIN wrote: Hi all, I would like to have your opinion about the best way to detect a asterisk failure, I mean when asterisk stop working but the process keep existing. There's a few ways you could do it. Something like: asterisk -rx 'iax2 show peers' | wc -l Would count the number of iax peers (assuming the command didn't return if asterisk wasn't working). Or you could connect to the manager interface on port 5038 and issue a few commands: http://www.voip-info.org/wiki-Asterisk+manager+API - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5bkxDQNt8rg0Kp4RAuTAAJ9fImZF2H1jfXz2uWvWD4xH9Or0vgCgohfc 5U2Sx92QzzACUOvawsE4eJw= =sGKd -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
Though still in the proof-of-concept stage, my project AstSee from http://www.astsee.com/ might be fun to play with if you're using linux/XWindows. There are screenshots there. Mojo satish patel wrote: Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Regards Looking for a deal? Find great prices on flights and hotels http://us.rd.yahoo.com/evt=47094/*http://farechase.yahoo.com/;_ylc=X3oDMTFicDJoNDllBF9TAzk3NDA3NTg5BHBvcwMxMwRzZWMDZ3JvdXBzBHNsawNlbWFpbC1uY20- with Yahoo! FareChase. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API - Originate command
Just to clear things up. It was one TCP connection to the manager interface and the originate commands are send in a batch. I was able to get away with 80 calls in a batch. Anything more than that is not good. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Sent: Monday, September 10, 2007 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Manager API - Originate command On 9/11/07, Wai Wu [EMAIL PROTECTED] wrote: Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. What did you mean by simultaneously? Opening 120 manager connections, and originating call at exactly the same time? I doubt.. So, probably there is some interval - within second/minute, etc.. And how many manager connections do you use? Maybe asterisk have some limit of them. Also - i think, there is some limit of asterisk accepting commands sequentially from one connection. Btw, what is your CPU load, when creating those 120 calls instantly? Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtptimeout on Asterisk 1.4.x
Hi Folks, Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed some dead calls apparently running for more than 8 hours. I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like this: chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61 seconds because it is directly bridged to another RTP stream I can kill that calls using 'soft hangup channel' but I'd like to know if its a new BUG introduced in 1.4.x releases and if possible, how to fix this? Thanks in advance. Rodrigo P. Telles ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API - Originate command
Thanks for sharing your experience. I will play around with the Asteirsk server tomorrow again. I took a look at it just before I left the office. It has loads of crap. It's got all those non-essential things and X windows running. Also, I can probably be able to get away with starting a call every 30-50ms. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, September 10, 2007 5:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Manager API - Originate command -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Wai Wu wrote: Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. First off, you should be using Async: true Secondly, you shouldn't really be doing 120 simultaneous calls. If your server can take say 300 concurrent calls, you will probably need to start those up with about 30ms between them. If you really need to start 120 calls all at the identical time, you probably want to be looking at clustering Asterisk servers. In SmoothTorque we set a minimum value for delay between calls, then have a funnel which accepts calls from predictive campaigns. The funnel knows about the connections to the Asterisk servers, and distributes the calls in a round robin fashion (assuming all servers are up). Each server has a queue which allows calls sent to that server to back up, and if a queue gets too full calls won't be sent to that server. If, after stopping the sending of calls to a server, the queue does not empty out, the server is placed into an inactive state, and a server marked as standby is moved into the active state. Any calls which remain in the queue are redistributed to other servers. Hope that helps! - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c T3+G284pc4LV/JMlj13v8gU= =oaJj -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API - Originate command
Just checked. I do have Async set to yes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Monday, September 10, 2007 7:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Manager API - Originate command Thanks for sharing your experience. I will play around with the Asteirsk server tomorrow again. I took a look at it just before I left the office. It has loads of crap. It's got all those non-essential things and X windows running. Also, I can probably be able to get away with starting a call every 30-50ms. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, September 10, 2007 5:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Manager API - Originate command -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Wai Wu wrote: Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. First off, you should be using Async: true Secondly, you shouldn't really be doing 120 simultaneous calls. If your server can take say 300 concurrent calls, you will probably need to start those up with about 30ms between them. If you really need to start 120 calls all at the identical time, you probably want to be looking at clustering Asterisk servers. In SmoothTorque we set a minimum value for delay between calls, then have a funnel which accepts calls from predictive campaigns. The funnel knows about the connections to the Asterisk servers, and distributes the calls in a round robin fashion (assuming all servers are up). Each server has a queue which allows calls sent to that server to back up, and if a queue gets too full calls won't be sent to that server. If, after stopping the sending of calls to a server, the queue does not empty out, the server is placed into an inactive state, and a server marked as standby is moved into the active state. Any calls which remain in the queue are redistributed to other servers. Hope that helps! - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG5bbeDQNt8rg0Kp4RArWFAKCoMPxaDmVLwPD+hupU9T8n+NuFYQCguq8c T3+G284pc4LV/JMlj13v8gU= =oaJj -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
Sir, I want to dedicate two or three Panasonic port to communicate with Asterisk and vise-versa. I am having Panasonic pbx 1232. Rajeev. Hello, 2007/9/10, C F [EMAIL PROTECTED]: Which Panasonic PBX? On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I am having Asterisk pbx which is running without any problem now i want to connect this with Panasonic pbx with FXS port so, if any body want to call panasonic users than he will call or vise-versa. How ? Do you plan to dedicate Panasonic PBX FXS ports to act as a trunk or would dedicate one port for each Asterisk user ? -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070910/1b2b2 6a8/attachment-0001.htm -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan Company, LLC wrote: Though still in the proof-of-concept stage, my project AstSee from http://www.astsee.com/ might be fun to play with if you're using linux/XWindows. There are screenshots there. that may be so, but without source, there's no way we can test it on freebsd. i'll stick with fop for the timebeing, thank you. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.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 | +=+ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
Hi, So, if you dedicate PBX ports to serve as a trunk, you're likely to loose the abilty to forward DID calls : when a call for an Asterisk user comes into Panasonic PBX, it will be forwarded to Panasonic FXS trunk ports. Then, Asterisk should have no mean to decode to which extension, the call has to be forwarded, has it comes from an FXO port which won't carry any data such as CallerID. I'm not 100% sure of that but that's the way analog ports works here, on some legacy PBX : analog port means no service. regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
