[asterisk-users] The problem DIAL with option T,t
HI This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext = 700 parkpos = 701-702 context = parkedcalls [featuremap] atxfer = *2 [applicationmap] set(DYNAMIC_FEATURES=tranf) tranf = *2,peer,waitexten(10|m) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Gtalk setup
Hi, I've just followed http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions from wiki, And i always get my jabber (GoogleTalk account for asterisk server) not registred: asterisk1*CLI jabber show connected Jabber Users and their status: User: myasteriskaccount - Disconnected asterisk1*CLI jabber test ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] deadalocks in asterisk
hi, i am recieving deadlocks frequently and its calls are getting hanged . Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! Aug 11 13:13:54 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! Aug 11 13:13:54 WARNING[6367] channel.c: Avoided initial deadlock for '0xb78e5be8', 10 retries! Aug 11 13:13:54 WARNING[6367] channel.c: Avoided initial deadlock for '0xb78e5be8', 10 retries! Aug 11 13:13:54 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! Aug 11 13:13:54 WARNING[6367] channel.c: Avoided initial deadlock for '0xb78e5be8', 10 retries! Aug 11 13:13:56 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! Aug 11 13:13:56 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! Aug 11 13:13:56 WARNING[6367] channel.c: Avoided initial deadlock for '0xb78e5be8', 10 retries! Aug 11 13:13:56 WARNING[6367] channel.c: Avoided initial deadlock for '0xb78e5be8', 10 retries! please let me know wat may be the problem and how to resolve it -- Thanks and Regards D.J.Sateesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DCAP on Linuxtag - Berlin 2009
Hello ! I am wondering if there is a possibility that Asterisk will be present on Linuxtag booth in Berlin 2009, and if there will be an option to take a DCAP exam ? Kind regards, Jan Prunk -- Jan Prunk janprunk AT SPAMFREE gmail DOT com Website: http://www.prunk.si PGP key: 00E80E86 Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Found unknown media description format
Hi One of my softphones is supposed to support g711 , however I am getting these errors and a 404 not found when I try to make a call from it. However on xlite it works fine using g711. Below is the log of the phone that is not working. Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0 a=ice-pwd:00gnam9Pd+SG6KzQNLf1fS a=ice-ufrag:Xng7 m=audio 19504 RTP/AVP 103 18 102 0 8 97 119 117 100 101 13 105 106 a=rtcp:19505 a=candidate:4 1 UDP 2122300927 192.168.0.176 19504 a=candidate:1 1 UDP 2122285311 169.254.2.2 19504 a=candidate:2 1 UDP 2122285055 192.168.238.1 19504 a=candidate:3 1 UDP 2122284799 192.168.111.1 19504 a=candidate:5 1 UDP 1694482431 193.227.186.146 19504 a=candidate:6 1 UDP 1677 87.236.144.70 41343 a=candidate:4 2 UDP 2122300926 192.168.0.176 19505 a=candidate:1 2 UDP 2122285310 169.254.2.2 19505 a=candidate:2 2 UDP 2122285054 192.168.238.1 19505 a=candidate:3 2 UDP 2122284798 192.168.111.1 19505 a=candidate:5 2 UDP 1694482430 193.227.186.146 19505 a=candidate:6 2 UDP 1676 87.236.144.70 41344 a=rtpmap:103 ISAC/16000 a=fmtp:18 annexb=no a=rtpmap:102 iLBC/8000 a=rtpmap:97 IPCMWB/16000 a=rtpmap:119 ISACLC/16000 a=rtpmap:117 red/8000 a=rtpmap:100 EG711U/8000 a=rtpmap:101 EG711A/8000 a=rtpmap:105 CN/16000 a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=sendrecv - --- (16 headers 33 lines) --- Sending to 87.236.144.9 : 5060 (no NAT) Using INVITE request as basis request - 0c49de60-1f17-4de8-aa0b-ae3f7b7527b9 Found no matching peer or user for 'xx.xx.xx.xx:5060' Found RTP audio format 103 Found RTP audio format 18 Found RTP audio format 102 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 119 Found RTP audio format 117 Found RTP audio format 100 Found RTP audio format 101 Found RTP audio format 13 Found RTP audio format 105 Found RTP audio format 106 Peer audio RTP is at port 192.168.0.176:19504 Found unknown media description format ISAC for ID 103 Found audio description format iLBC for ID 102 Found unknown media description format IPCMWB for ID 97 Found unknown media description format ISACLC for ID 119 Found unknown media description format red for ID 117 Found unknown media description format EG711U for ID 100 Found unknown media description format EG711A for ID 101 Found audio description format CN for ID 105 Found audio description format telephone-event for ID 106 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.176:19504 Looking for 5678 in default (domain ser..net) Any ideas ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 SVN / dahdi / meetme / - unable to open pseudo device
Hi, I was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [EMAIL PROTECTED]:1] MeetMe(SCCP/6000-0001, 444|dcIM) in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775 build_conf: Unable to open pseudo device [Aug 11 14:04:45] -- SCCP/6000-0001 Playing 'conf-invalid' (language 'en') [Aug 11 14:04:49] == Spawn extension (client_int_sgmobile, 8001, 1) exited non-zero on 'SCCP/6000-0001' Asterisk SVN-branch-1.4-r137138 Funnily, 1.6-trunk works with that dahdi version... Any ideas? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deadalocks in asterisk
D.J.Sateesh [EMAIL PROTECTED] writes: hi, i am recieving deadlocks frequently and its calls are getting hanged . Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! We upgrade customers who hit that bug to 1.4... The locking is greatly improved. Which reminds me, is there an easy way to see what Asterisk uses memory for? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scala and Asterisk-Java (was RE: Auto Dialer proof of concept)
Here's my attempt to explain a quick way of doing an auto dialer with Scala and the Asterisk-Java library: http://blogs.reucon.com/asterisk-java/2008/08/10/outbound_message_delive ry_using_agi_and_ami_in_scala.html Cheers, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brad Sent: Friday, August 08, 2008 4:28 PM To: emist Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Auto Dialer proof of concept Yes, everyone will have the same message. You think building the call fill in the spooler is the most effectient? Can you refer me to a page that will explain pulling the info from a sql db into a call file? Last thing, I dial out to an extension, not a registered sip provider, my provider does not require authentication. How would I pull from DB, put into call file, send to context. Short and pretty. Just trying to get my head back into Asterisk. --- On Fri, 8/8/08, emist [EMAIL PROTECTED] wrote: From: emist [EMAIL PROTECTED] Subject: Re: [asterisk-users] Auto Dialer proof of concept To: [EMAIL PROTECTED] Date: Friday, August 8, 2008, 4:18 PM Hey Brad, The simplest way I thought to implement it for a client who needed multiple calls to be placed based on time was to code a deamon that would query the db every given interval, check if there were any calls that needed to be made and pull those out and build call files. That seemed to work pretty decent. However, if you just want to call 2000 people with the same message with the click of a button all you'd have to do is have a frontend that would pull the message off the db along with each person's number and build call files in a loop. Thats simple and relatively scalable, unless you're doing 1,000,000 at a time or something. Regards, Igor H. Brad wrote: I read that last night and I was curious about followme' Will this give me the ability to dial out 10 - 2000 simultaneously calls the easiest and control to number of call? doing it the file method looks kind of easy for proof of concept, but not very manageable. I could seeing putting 2000 files into a directory would be very cumbersome. Eventually, the number will be coming from a sql database. I am just trying to get general concept to prove it works for now, but do not want to have to reconfigure to much over the weekend --- On Fri, 8/8/08, emist [EMAIL PROTECTED] wrote: From: emist [EMAIL PROTECTED] Subject: Re: [asterisk-users] Auto Dialer proof of concept To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, August 8, 2008, 3:57 PM http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Bradley Sumrall wrote: I am a returning Asterisk user. It has been a few years since I played with it and trying to get a server up for proof of concept What is the easiest method of having asterisk dial 5 numbers simultainiously and deliver a pre recorded message? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 SVN / dahdi / meetme / - unable to open pseudo device
Stefan Gofferje wrote: Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [EMAIL PROTECTED]:1] MeetMe(SCCP/6000-0001, 444|dcIM) in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775 build_conf: Unable to open pseudo device [Aug 11 14:04:45] -- SCCP/6000-0001 Playing 'conf-invalid' (language 'en') [Aug 11 14:04:49] == Spawn extension (client_int_sgmobile, 8001, 1) exited non-zero on 'SCCP/6000-0001' Asterisk SVN-branch-1.4-r137138 Fixed in revision 137188; this module apparently did not get any DAHDI conversion work at all, but I don't know how it got missed. Thanks for the testing! -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime Unregister
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11 21:37:31] -- Called 102104
until it reached the timed out i set in the dialplan which is 30 secs
Dial(SIP/${EXTEN}|30|t|M(setmusiconhold,moh-${EXTEN}))
is there something i can add on my dialplan to first detect that the
user is not available, or maybe force unregister, anything that would
not make my dialplan to wait for 30 secs.
also i'm not using rtcachefriends, how would i know in the CLI which
user is registered? i tried sip prune but it shows me nothing
sip prune realtime peer all
No peers found to prune.
anyone experienced this?
thank you
regards.
nhadie
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Re: [asterisk-users] DCAP on Linuxtag - Berlin 2009
On Mon, 2008-08-11 at 11:20 +0200, Jan Prunk wrote: I am wondering if there is a possibility that Asterisk will be present on Linuxtag booth in Berlin 2009, and if there will be an option to take a DCAP exam ? While Digium typically sends a few people to present at the Linuxtag show every year, I don't think we've ever offered the dCAP exam at Linuxtag. I don't yet have our training classes scheduled that far in advanced, but I'll certainly keep it in mind as we set next year's training schedule. As an alternative, you can also take the dCAP on the last day of any of our regularly scheduled Asterisk Bootcamp or Asterisk Advanced training classes in your area. Please check our website for more information on upcoming training classes around the world. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The problem DIAL with option T,t
Hi Larry - This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext = 700 parkpos = 701-702 context = parkedcalls [featuremap] atxfer = *2 [applicationmap] set(DYNAMIC_FEATURES=tranf) tranf = *2,peer,waitexten(10|m) You've got a few problems here: 1) You have two different operations set to: *2 You can only have one feature per key combination 2) You can't set the DYNAMIC_FEATURES variable in the features.conf file. You can only set variables in extensions.conf (or extensions.ael) 3) If you just need to set up attended transfer, you only need the line atxfer = *2 and nothing else. Attended transfer is a pre-defined feature. The [applicationmap] section is for creating new features that aren't pre-defined. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 SVN / dahdi / meetme / - unable to open pseudo device
Kevin P. Fleming schrieb: Fixed in revision 137188; this module apparently did not get any DAHDI conversion work at all, but I don't know how it got missed. Thanks for the testing! Confirmed. Works fine now under all (extensively) tested conditions. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] out going call files and correct dial status
Hi all, I am using outgoing call files to place calls. Issue is when that call is BUSY I dont get the correct DIALSTATUS from that call when running my AGI and the failed extension. WHERE can I make a change in the code so that the DIALSTATUS when the call ended can be added as a variable in my call file (not sure why this isnt there already) - so when my AGI runs based on the failed extension I can do a GET VARIABLE inquiry from the call file and get that status. Thanks for any suggestions / pointers... Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Unregister
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
See http://www.voip-info.org/wiki/view/Asterisk+sip+qualify for more
information. Short version is that configuring a qualify interval is
the equivalent of setting up a heartbeat between Asterisk and registered
devices configured with a qualify interval. If the heartbeat fails, the
phone's registration is suspended.
Nhadie wrote:
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11 21:37:31] -- Called 102104
until it reached the timed out i set in the dialplan which is 30 secs
Dial(SIP/${EXTEN}|30|t|M(setmusiconhold,moh-${EXTEN}))
is there something i can add on my dialplan to first detect that the
user is not available, or maybe force unregister, anything that would
not make my dialplan to wait for 30 secs.
also i'm not using rtcachefriends, how would i know in the CLI which
user is registered? i tried sip prune but it shows me nothing
sip prune realtime peer all
No peers found to prune.
anyone experienced this?
thank you
regards.
nhadie
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Re: [asterisk-users] Asterisk Gtalk setup
Works ok on my side. Any debug messages from your console you could post here? Thanks, Philippe On Mon, Aug 11, 2008 at 9:16 AM, vivek rastogi [EMAIL PROTECTED] wrote: Hi, I've just followed http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions from wiki, And i always get my jabber (GoogleTalk account for asterisk server) not registred: asterisk1*CLI jabber show connected Jabber Users and their status: User: myasteriskaccount - Disconnected asterisk1*CLI jabber test ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Unregister
Thank you for your reply sir. I tried setting qualify=yes my CPU spiked
to 113%
i continuously see this on my CLI
Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (388ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '110100' is now Lagged. (2354ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118777' is now Reachable. (432ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118777' is now Reachable. (436ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (440ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (444ms / 2000ms)
could that be the cause of high cpu? i'm logged in on the cli asterisk
-vr, my verbosity is only set to 1. how come i keep on seeing the NOTICE?
thanks again in advanced
regards,
nhadie
Rob Hillis wrote:
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
See http://www.voip-info.org/wiki/view/Asterisk+sip+qualify for more
information. Short version is that configuring a qualify interval is
the equivalent of setting up a heartbeat between Asterisk and registered
devices configured with a qualify interval. If the heartbeat fails, the
phone's registration is suspended.
Nhadie wrote:
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11 21:37:31] -- Called 102104
until it reached the timed out i set in the dialplan which is 30 secs
Dial(SIP/${EXTEN}|30|t|M(setmusiconhold,moh-${EXTEN}))
is there something i can add on my dialplan to first detect that the
user is not available, or maybe force unregister, anything that would
not make my dialplan to wait for 30 secs.
also i'm not using rtcachefriends, how would i know in the CLI which
user is registered? i tried sip prune but it shows me nothing
sip prune realtime peer all
No peers found to prune.
anyone experienced this?
thank you
regards.
nhadie
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Re: [asterisk-users] deadalocks in asterisk
Benny Amorsen wrote: D.J.Sateesh [EMAIL PROTECTED] writes: hi, i am recieving deadlocks frequently and its calls are getting hanged . Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for '0xb6692258', 10 retries! That is actually more of a debug message, and is not necessarily an indication of a problem. We upgrade customers who hit that bug to 1.4... The locking is greatly improved. Which reminds me, is there an easy way to see what Asterisk uses memory for? Yeah. Compile Asterisk with the MALLOC_DEBUG option. With that, you will have the CLI commands memory show summary and memory show allocations, which show you all of the memory allocations that Asterisk has made by file, function, and line number. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP TLS error: ast_make_file_from_fd: FILE * open failed
Stefan Gofferje wrote: [Aug 8 23:30:13] SSL certificate ok [Aug 8 23:30:13] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd: FILE * open failed! First, try the latest code in the Asterisk 1.6.0 branch. Also, make sure you're using a reasonably current version of OpenSSL. $ svn co http://svn.digium.com/svn/asterisk/branches/1.6.0 If you still have trouble, feel free to report it on http://bugs.digium.com/. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Stefan Gofferje wrote: I have configured all IAX clients with encryption. I use Zoiper as a softphone. When I make a call in the LAN from desktop-PC to *, the call is - according to wireshark not encrypted. Wireshark identifies the packets as normal G.711 mu-law packets. However, * reports the client as encrypted: k-tanco*CLI iax2 show peers Name/UsernameHost Mask Port Status sgofferj RFC-1918 IP(D) 255.255.255.255 4570 (E) OK (2 ms) Funnily, if my friend calls me from internet - also with Zoiper - Wireshark cannot identify the packets so I conclude, the call is encrypted. Does this make any sense? You'd have to provide a packet capture to see exactly what is happening. It sounds like on the call leg between your client and Asterisk, it isn't offering encryption as a capability, so it doesn't get used. However, when your friend calls you, and Asterisk makes a call out to your client, it offers encryption, and your client accepts it. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate Status Monitoring
Hi all, I'm writing an application to Queue and Manage AMI Originate actions. Basically, callfiles on steroids if you may :) I'm facing the following challenges, and any ideas or pointers will be hugely appreciated. 1. When I successfully Queue an Originate... (Response is Success), how will i know when the channel has been freed up? I had initially been monitoring the Response/Message/ActionID trio, but i discovered that this returns immedietly asterisk queues the command. I've also noticed that Events keep popping up during the life-time of the Origination. I'm guessingthe solution would be to monitor for a particular event. My question then is... can i _always_ count on Even: Hangup, always being present, _nomatter_ what? (As i've noticed Hangup is raised even on failed events) 2. I have noticed that in version 1.4.21, the events do not carry any associated ActionID (which is understandable, as they may not be in response to any AMI commands). My question then i how can i reliably detect which events have to do with me? If i have a channel SIP/foo, i notice that the events carry a channel tag: SIP/foo-xxx, where xxx is a hex number that I don't know how to generate, so if i have to channels, SIP/foo and SIP/foo1, it gets tricky knowing which events belong to which channel. Especially when trying to multiplex just one socket connection. 3. Uhh there is really no #3 :), but any other tips or problems hints at solutions to problems I *may* face, from experience, are very much welcome. Thnx. cheers, Essien OS: Linux 2.6.25 Asterisk: 1.4.21 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 25
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk broadcast to web
Hi all, I have an interesting problem that I am looking for a solution for. I want to be able to call into an asterisk server and have what I say be broatcast over a streaming web radio station. I imagine using something like icecast for that. Does anybody have any pointers on how to get started? I am stuck on how to get the audio out of asterisk to be able to put into something like icecast. Any help or suggestions would be appreciated. Thanks, _ /-\ ndrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk broadcast to web
Off the top of my head... you could probably route the audio of a softphone (like Zoiper/X-Lite) to something like Nicecast (Mac) or Icecast. On 11 Aug, 2008, at 9:21 AM, Andrew Niemantsverdriet wrote: Hi all, I have an interesting problem that I am looking for a solution for. I want to be able to call into an asterisk server and have what I say be broatcast over a streaming web radio station. I imagine using something like icecast for that. Does anybody have any pointers on how to get started? I am stuck on how to get the audio out of asterisk to be able to put into something like icecast. Any help or suggestions would be appreciated. Thanks, _ /-\ ndrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate Status Monitoring
Hi, The only reliable solution I've found for this is to set a custom variable with the Originate action and query new channels for that variable when they appear. We've also used this strategy successfully when implementing Asterisk-Java's live API. Depending on which language you are going to use for your application solutions similar to Asterisk-Java may be available that hide this and similar obstacles. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate Status Monitoring
Essien Ita Essien wrote: Hi all, I'm writing an application to Queue and Manage AMI Originate actions. Basically, callfiles on steroids if you may :) I'm facing the following challenges, and any ideas or pointers will be hugely appreciated. 1. When I successfully Queue an Originate... (Response is Success), how will i know when the channel has been freed up? I had initially been monitoring the Response/Message/ActionID trio, but i discovered that this returns immedietly asterisk queues the command. You have to mod the Event: Hangup manager event to include something useful for you. This is an example of adding the CallerIDNum/Name to the event. http://dynx.net/ASTERISK/gnudialer/channel.c.hangup_callerid_ast1419.diff I've also noticed that Events keep popping up during the life-time of the Origination. I'm guessingthe solution would be to monitor for a particular event. My question then is... can i _always_ count on Even: Hangup, always being present, _nomatter_ what? (As i've noticed Hangup is raised even on failed events) On failed 'attempts' it would be an origination 'failure', and those you need to use a... ; this extension MUST be here for OriginateFailure triggers exten = failed,1,Hangup in the context you use for your originate attempt. That will produce another manager event, which will provide a Reason: code. That code is the 'AST_CONTROL' of what last happened on that channel. frame.h:#define AST_CONTROL_HANGUP 1 frame.h:#define AST_CONTROL_RING2 frame.h:#define AST_CONTROL_RINGING 3 frame.h:#define AST_CONTROL_ANSWER 4 frame.h:#define AST_CONTROL_BUSY5 frame.h:#define AST_CONTROL_TAKEOFFHOOK 6 frame.h:#define AST_CONTROL_OFFHOOK 7 frame.h:#define AST_CONTROL_CONGESTION 8 ... 2. I have noticed that in version 1.4.21, the events do not carry any associated ActionID (which is understandable, as they may not be in response to any AMI commands). My question then i how can i reliably detect which events have to do with me? Refer to the calleridnum/name section above. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Russell Bryant schrieb: You'd have to provide a packet capture to see exactly what is happening. It sounds like on the call leg between your client and Asterisk, it isn't offering encryption as a capability, so it doesn't get used. However, when your friend calls you, and Asterisk makes a call out to your client, it offers encryption, and your client accepts it. Hm, not sure if I get your point. This is the infrastructure (exempt): Zoiper --LAN-- Asterisk --INET-- Zoiper (my) | (friend) | Cisco phone When I dial the Cisco phone from my Zoiper, wireshark shows unencrypted packets. When my friend calls the Cisco phone from her Zoiper, wireshark shows unknown = encrypted(?) packets. We are both using the same Zoiper release, just she on MAC and I on Windows PC. I also now tested to make a call from the Cisco phone to my Zoiper - also no encryption. Would it make sense to introduce a parameter forceencryption=yes per peer in iax.conf? In sensitive environments, people want to be certain that a call is encrypted. They probably rather want a call to fail than have a call that might be unencrypted without knowing it. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile: scrambled audio, no MOH, no call signalization
This is how it sounds: http://stefan.gofferje.net/chan_mobile_distorted.wav Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk broadcast to web
If you want inbound speech then you can use one of the web based softphones (mexuar etc), just through them into a conference room with all but the 'moderators' into a silenced one way conference room and use DTMF to raise hands etc. But for bandwidth capacity issues you can use something as simple as a jerry rigged BT101 handset with a speaker connected to a microphone socket of a pc running icecast which will give you one way speech. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Niemantsverdriet Sent: Monday, 11 August 2008 12:21 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk broadcast to web Hi all, I have an interesting problem that I am looking for a solution for. I want to be able to call into an asterisk server and have what I say be broatcast over a streaming web radio station. I imagine using something like icecast for that. Does anybody have any pointers on how to get started? I am stuck on how to get the audio out of asterisk to be able to put into something like icecast. Any help or suggestions would be appreciated. Thanks, _ /-\ ndrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Asterisk out of the RTP media path
SIP wrote: When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP media path. I understand that this is sort of the idea behind a bridged channel, but is there any way to avoid it? Is there any way to say Connect this number and this number and then get out of the way, or is this a design limitation? N. No ideas on this one? I've tried everything I can think of and then some and still can't get Asterisk out of the media path. I can do it if I don't originate the call with Asterisk, but only use Asterisk to connect one leg of the call, but if I use Asterisk to connect both legs, no luck. Going about this the wrong way? N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime CLI command
Hi, Are there Asterisk CLI commands I can use to add/manage extensions and dial plans when using Asterisk Realtime with MySQL? I know about the database put and database get commands, but from what I've read they apply to AstDB and not to the Asterisk Realtime database. So far I've been manually INSERTing and UPDATEing the records in MySQL. Thanks jm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 26
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Michael Graves schrieb: Which Asterisk systems provide automatic provisioning of phones? Gemeinschaft ( http://www.amooma.de/gemeinschaft/ ) does. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Russell Bryant schrieb: Interesting. Here are a couple more sanity checks you can do. First, double check to ensure that your entry in iax.conf has encryption=yes set. Also, when you make the call into Asterisk, set the verbose setting up a bit. You should see output from chan_iax2 which indicates what peer you are authenticating as. Make sure that the call is matching the entry that you think it is. I will do some more testing as you suggested. Also, is there any encryption option in Zoiper that you have to enable? Not to my knowledge. I will send an issue report to asteriskguru also. Would it make sense to introduce a parameter forceencryption=yes per peer in iax.conf? In sensitive environments, people want to be certain that a call is encrypted. They probably rather want a call to fail than have a call that might be unencrypted without knowing it. That is a good suggestion. Opened a bug for that (0013285) :). Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime CLI command
On Monday 11 August 2008 12:07:47 J.M. wrote: Are there Asterisk CLI commands I can use to add/manage extensions and dial plans when using Asterisk Realtime with MySQL? I know about the database put and database get commands, but from what I've read they apply to AstDB and not to the Asterisk Realtime database. So far I've been manually INSERTing and UPDATEing the records in MySQL. There's no such command, no. I'm not sure it would even be a good idea, given the sheer number of fields and the fact that there are much better interfaces to the database (even the mysql CLI is better). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 variable sharing
Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] manager/originate
Robor Oghene schrieb: Please let someone throw more light on this command and it usage.. i tried a search but can't to get anything useful. asterisk -rx 'manager show command Originate' Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 variable sharing
Ruddy Gbaguidi wrote: Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks Back in 1.2 you had to use type 'friend' to pass vars, as the 'peer' structure didn't have vars, only 'user' did. It is/was probably the same with 1.4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone system layout suggestions
I am thinking about a change to our company's phone layout and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to the headquarters via a toll free number. We are adding another remote office and are planning a MPLS network to connect all locations for DATA. As we really want a centralized switchboard anyway, I thought with QoS and an MPLS, I could eliminate the remote PBXs (and not have to buy another for the new locations) by simply using the MPLS to tie everything together. See below. /\ | CORP HEADQUARTERS | PRI (23 Channels, 100 DIDs) | Asterisk PBX | | 30 Polycom 501s | \/ | | 3Mbps MPLS (2 T1s) | /\ | MPLS Cloud | \/ | | | | | \---SIP over MPLS 1.5Mbps T1---Branch Office 1 (5 Polycom 501s) | | | \---SIP over MPLS 1.5Mbps T1---Branch Office 2 (5 Polycom 501s) | \---SIP over MPLS 1.5Mpbs T1---Branch Office 3 (3 Polycom 501s) Question 1: I've never had an MPLS network with QoS. Will the call quality Really be as good as ATT assures me it will be with QoS? Assuming we never have more than 5 calls going over the MPS from a branch? Question 2: MPLS are pretty reliable, but last mile connections can be cut, hardware can fail at ATT or whatever. Is this putting too many eggs in one basket? If I lose the headquarters T1s (MPLS or PRI), everyone is down. Would You do it this way? Question 3: (OT) For those who have used an MPLS. How much better throughput for DATA (NOT VoIP) should I see compared to using the Internet? I'm mostly just curious here and realize it is hard to compare, but when I do any type of file transfer between office right now, I use FTP over the Internet and both ends have a T1. Assuming an MPLS on each end, what is your experience when comparing average throughput compared to an Internet transfer. Just a guess of what you've seen. (i.e. Yes, you'll see a big difference, maybe a little better or couldn't really see that big of a difference) Thanks for your input. Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HP server and Meetme applications
Hi list I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ? and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? AyMaN ALMONTAHA .ICT 11 AUG 2008 _ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 variable sharing
It doesn't seems to be working ... What I wanted to do is on the first server, Set a channel variable... then dial the number. When I received the call on the remote server, use that variable ... Is it possible ? Richard Lyman wrote: Ruddy Gbaguidi wrote: Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks Back in 1.2 you had to use type 'friend' to pass vars, as the 'peer' structure didn't have vars, only 'user' did. It is/was probably the same with 1.4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
aymen warfalli wrote: Hi list I got one *HP* ProLiant *DL380 G5* - *Quad*-*Core* Xeon E5440 2.83 with 4 gig *RAM* I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ?and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? AyMaN ALMONTAHA .ICT 11 AUG 2008 Whatever answer you get, I would approach this project in a SLOW, METHODICAL manner. i.e put 1 E1 card, get system performance metrics and user experience ADD the second card, TEST for voice quality and gather metrics. And then TEST some more. It will Likely work, BUT, I think you are venturing into an area with some large potential alligators. But, don't plug everything in Friday afternoon and expect zero problems Monday morning :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 variable sharing
TP'n to follow broken flow. As i stated, you must use a 'user to user' (friend) as the iax2_user structure has struct ast_variable *vars, the iax2_peer (and iax2_trunk_peer) do NOT. Therefore you cannot pass *channel variables* when using peer-user setups, only user-user setups. Which means, you must setup both sides as user (which 'friend' on both sides will do). There are associated security issues with this setup (which is why is it not advised), especially when you do not fully understand this interaction. (which you appear to not understand) I have visually confirmed this is still true in up to asterisk 1.4.19 Ruddy Gbaguidi wrote: It doesn't seems to be working ... What I wanted to do is on the first server, Set a channel variable... then dial the number. When I received the call on the remote server, use that variable ... Is it possible ? Richard Lyman wrote: Ruddy Gbaguidi wrote: Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks Back in 1.2 you had to use type 'friend' to pass vars, as the 'peer' structure didn't have vars, only 'user' did. It is/was probably the same with 1.4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 variable sharing
Erm, I've been out of the loop, but in 1.6 there's the IAXVAR dialplan function that does _exactly_ what you want. I don't know if it's been backported to 1.4, but I think there was a patch at one point. Tim. On 11 Aug 2008, at 20:43, Richard Lyman wrote: TP'n to follow broken flow. As i stated, you must use a 'user to user' (friend) as the iax2_user structure has struct ast_variable *vars, the iax2_peer (and iax2_trunk_peer) do NOT. Therefore you cannot pass *channel variables* when using peer-user setups, only user-user setups. Which means, you must setup both sides as user (which 'friend' on both sides will do). There are associated security issues with this setup (which is why is it not advised), especially when you do not fully understand this interaction. (which you appear to not understand) I have visually confirmed this is still true in up to asterisk 1.4.19 Ruddy Gbaguidi wrote: It doesn't seems to be working ... What I wanted to do is on the first server, Set a channel variable... then dial the number. When I received the call on the remote server, use that variable ... Is it possible ? Richard Lyman wrote: Ruddy Gbaguidi wrote: Hi all Back in the 1.2 days I think, there were some discussions about how two asterisk servers can share channel variables through an IAX protocol. I don't see anything in 1.4 at least to be able to make it done. Thanks Back in 1.2 you had to use type 'friend' to pass vars, as the 'peer' structure didn't have vars, only 'user' did. It is/was probably the same with 1.4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phone rings once before playing message
My phone rings once and stops before playing message; how to stop this behavior. Could it have something to do with this error: channel_find_locked: Avoided initial deadlock for '0x81c04d0', 9 retries! Here is the dial plan: exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,SetMusicOnHold(default) exten = s,4,Background(office-closed) exten = s,5,Voicemail(441) exten = s,6,Hangup() -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
On Mon, Aug 11, 2008 at 02:45:18PM -0400, aymen warfalli wrote: Hi list Hi. Please don't thread-jack. I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 I would be surprised. I have a Core2Quad 2.4G machine with one Sangoma quad-T card (admittedly, with*out* HWEC), that's meetme-ing 72 Zap channels from Channelbanks into an equivalent number of IAX channels over 100BaseT; each conference has at least one extra Record() app running full time, as well as one playback. This is a production machine, but it takes careful tuning to keep it production-stable at that load (where my definition of production-stable is if the load average breaks 2.0 for more than 5 consecutive half-minutes, I get nervous.). You *might* get that to work, with HWEC, but I have no experience with whether MeetMe is going to behave if you want it all to be One Big Conference -- it will *certainly* depend on which Asterisk release you're running, and my intuition says if you have to ask, you're not the guy to set it up; pay someone. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing call file and agi detect busy
Jerry Geis wrote: Call files spawn a completely new channel that your AGI most likely isn't going to be able to track. Since the call is a completely new channel, the DIALSTATUS variable for this channel will not be visible to your original channel. You may want to look at using the Originate action from within the manager API. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate So there is no way either in the dialplan or in the AGI that I can find out a status of my call? As to why it did or did not complete? Doesnt that seem like a defect? There has to be a way around that? I dont see off hand how this manager originate is any different. Jerry After searching the code FAR and WIDE it comes down to the SIMPLE result of not looking at DIALSTATUS in this case but looking at REASON. REASON is set to 5 - busy and everything is good. So I was just looking at the wrong environment variable. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phone rings once before playing message
On 08/11/08 14:38, Joseph wrote: My phone rings once and stops before playing message; how to stop this behavior. I think it has something to do with Linksys SPA 3201 with Setting under: PSTN-To-VoIP Gateway. PSTN-To-VoIP Gateway Enable: Yes PSTN Ring Thru Line 1: Yes PSTN Answer Delay: 3 (this is enough time to pass caller ID) With the current setting the phone YES, YES, the ring on line1 rings one and stops With setting YES, NO the line1 does not ring. -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
On 8/11/08, aymen warfalli [EMAIL PROTECTED] wrote: I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ? and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? I have set up a system with 180 users in meetme rooms on a single server (4 x dual core Xeons) using a Sangoma a108D(8 port T1/E1 card with hardware EC with 8 x T1s connected) and the machine was running at high load but it was usable with good audio. Not sure what adding another 60 channels to that would do in terms of load or audio quality. What is the exact application you are trying to build? What capacity does the meetme room need to have in total? I have actually built distributed meetme applications where you have multiple servers that you can connect meetme rooms on one server to another and have essentially unlimited capacity in a single functional conference room as long as you have the hardware for it. Shameless plug I am going to be talking about this very subject at Astricon next month, along with 2 other presentations I'm giving there, if you happen to be going. /Shameless plug MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
On 8/11/08, aymen warfalli [EMAIL PROTECTED] wrote: I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ? and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? I have set up a system with 180 users in meetme rooms on a single server (4 x dual core Xeons) using a Sangoma a108D(8 port T1/E1 card with hardware EC with 8 x T1s connected) and the machine was running at high load but it was usable with good audio. Not sure what adding another 60 channels to that would do in terms of load or audio quality. What is the exact application you are trying to build? What capacity does the meetme room need to have in total? I have actually built distributed meetme applications where you have multiple servers that you can connect meetme rooms on one server to another and have essentially unlimited capacity in a single functional conference room as long as you have the hardware for it. Shameless plug I am going to be talking about this very subject at Astricon next month, along with 2 other presentations I'm giving there, if you happen to be going. /Shameless plug MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Unregister
I disabled logging of NOTIFY on the CLI and it does not show anymore,
however CPU is still very high, latency as well goes up when it is
trying to poke my phone here, my phone(SPA942) also keeps on rebooting
is there a way to increase the time of sending the qualify? TIA
regards
nhadie
Nhadie wrote:
Thank you for your reply sir. I tried setting qualify=yes my CPU spiked
to 113%
i continuously see this on my CLI
Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (388ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '110100' is now Lagged. (2354ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118777' is now Reachable. (432ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118777' is now Reachable. (436ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (440ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (444ms / 2000ms)
could that be the cause of high cpu? i'm logged in on the cli asterisk
-vr, my verbosity is only set to 1. how come i keep on seeing the NOTICE?
thanks again in advanced
regards,
nhadie
Rob Hillis wrote:
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
See http://www.voip-info.org/wiki/view/Asterisk+sip+qualify for more
information. Short version is that configuring a qualify interval is
the equivalent of setting up a heartbeat between Asterisk and registered
devices configured with a qualify interval. If the heartbeat fails, the
phone's registration is suspended.
Nhadie wrote:
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11 21:37:31] -- Called 102104
until it reached the timed out i set in the dialplan which is 30 secs
Dial(SIP/${EXTEN}|30|t|M(setmusiconhold,moh-${EXTEN}))
is there something i can add on my dialplan to first detect that the
user is not available, or maybe force unregister, anything that would
not make my dialplan to wait for 30 secs.
also i'm not using rtcachefriends, how would i know in the CLI which
user is registered? i tried sip prune but it shows me nothing
sip prune realtime peer all
No peers found to prune.
anyone experienced this?
thank you
regards.
nhadie
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[asterisk-users] Intermittent T.38 pass through
Hi All, I've been testing reliability with t.38 faxing pass through with * 1.4.21.1, Linksys ATA's 2102 x 2, Sharp UX-B800SE and Cannon ImageClass D880. cannon 2102 #1 SIP * SIP 2102 #2 sharp Started out with default settings on all devices, configured Asterisk to handle T.38 pass through, the configuration I believe is solid. I get relaiable results faxing from the sharp to the cannon. I get intermittent results, 50% success or failure, faxing from cannon to sharp. The first thing I did was to switch the faxes, so the ATA's and Asterisk remain the same, just switched the faxes to the oposite end of the path. Doing this comfirmed the original results, sharp to cannon is reliable, cannon to sharp is unreliable. What I observe faxing from sharp to cannon: path sets up as ulaw RTP, cannon answers, RTP switched to UDPTL, fax completes Faxing from cannon to sharp: path sets up as ulaw RTP, sharp answers, RTP switches to UDPTL only half the time when UDPTL is active, fax completes, when the path stays with RTP, fax always fails If I understand correctly, Asterisk switches the media stream to UDPTL when it hears valid fax tones on each side of the path, if it only detects fax tones on 1 path leg, then it keeps the media path through RTP. Or is the mechanism switching to UDPTL in the SIP headers? So, I adjusted db levels on the FXS ports, higher and lower, no effect. I increased jitter, reduced jitter, disabled jitter, no effect. Ensured echo can's were off, no effect. Manually set faxes to 14.4bps, ecm off, no effect. Even switched telephone cord, no effect. On these Linksys 2102's, you can predial #99 to force the ATA to enable fax t.38, this works and is reliable, no RTP is setup, just UDPTL. So my question is this: Can I setup Asterisk to only allow t.38 pass through from these ATA's, without the need to use the #99 in every dial string from the fax machine? Thanks. JR -- - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can I get a little criticism?
dear list
I'm very new in telephony and asterisk especial. so, it would be great
if somebody see at my dialplan. it works (except e1 which is untested at
this day), but I think it's not perfect. thanks
alexander
p.s. in Russia national prefix is 8 and international 810
/etc/asterisk/dialplan-
[phones]
;send to westcall e1
exten = _89.,1,Goto(westcall,${EXTEN},1) ;spb cellular like 911, 921 etc.
exten = _ZXX,1,Goto(westcall,${EXTEN},1) ;ptsn numbers
exten = _00XX,1,Goto(westcall,${EXTEN:2},1) ;01, 02, 03 etc.
;send to sipnet
exten = _8N.,1,Goto(sipnet,7${EXTEN:1},1) ;national
exten = _810X.,1,Goto(sipnet,${EXTEN:3},1) ;international
;send to internal
exten = _01XX,1,Goto(internal,${EXTEN},1) ;internal numbers
[internal]
;0100 fxs1
exten = 0100,1,Dial(ZAP/32,30,rtT)
exten = 0100,n,Hangup()
;0101 sip
exten = 0101,1,Dial(SIP/0101,30,rtT)
exten = 0101,n,Hangup()
;0102 sip
exten = 0102,1,Dial(SIP/0102,30,rtT)
exten = 0102,n,Hangup()
;0103 sip
exten = 0103,1,Dial(SIP/0103,30,rtT)
exten = 0103,n,Hangup()
;0104 sip
exten = 0104,1,Dial(SIP/0104,30,rtT)
exten = 0104,n,Hangup()
;0105 sip
exten = 0105,1,Dial(SIP/0105,30,rtT)
exten = 0105,n,Hangup()
[westcall]
;try channels from 1 to 10 or hang
exten = _X.,1,Set(CHAN=1)
exten = _X.,n(trychan),Dial(ZAP/${CHAN}/${EXTEN})
exten = _X.,n,Set(CHAN=$[${CHAN} + 1])
exten = _X.,n,GotoIf($[${CHAN} = 10]?trychan:bye)
exten = _X.,n(bye),Congestion()
exten = _X.,n,Hangup()
[sipnet]
;see users.conf
exten = _Z.,1,Dial(SIP/sipnet/${EXTEN})
[incoming]
;actually only e1 incomings
exten = s,1,Queue(mainq,rtT) ;strategy = roundrobin
exten = s,n,Congestion()
exten = s,n,Hangup()
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Re: [asterisk-users] phone rings once before playing message
Joseph wrote: On 08/11/08 14:38, Joseph wrote: My phone rings once and stops before playing message; how to stop this behavior. I think it has something to do with Linksys SPA 3201 with Setting under: PSTN-To-VoIP Gateway. PSTN-To-VoIP Gateway Enable: Yes PSTN Ring Thru Line 1: Yes PSTN Answer Delay: 3 (this is enough time to pass caller ID) With the current setting the phone YES, YES, the ring on line1 rings one and stops With setting YES, NO the line1 does not ring. When playing with PSTN equipment, it's _very_ hard to get rid of the rings. And in some countries it's actually illegal (I don't know why, but I know it is) PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone system layout suggestions
Bill Andersen wrote: I am thinking about a change to our company's phone layout and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to the headquarters via a toll free number. We are adding another remote office and are planning a MPLS network to connect all locations for DATA. As we really want a centralized switchboard anyway, I thought with QoS and an MPLS, I could eliminate the remote PBXs (and not have to buy another for the new locations) by simply using the MPLS to tie everything together. See below. /\ | CORP HEADQUARTERS | PRI (23 Channels, 100 DIDs) | Asterisk PBX | | 30 Polycom 501s | \/ | | 3Mbps MPLS (2 T1s) | /\ | MPLS Cloud | \/ | | | | | \---SIP over MPLS 1.5Mbps T1---Branch Office 1 (5 Polycom 501s) | | | \---SIP over MPLS 1.5Mbps T1---Branch Office 2 (5 Polycom 501s) | \---SIP over MPLS 1.5Mpbs T1---Branch Office 3 (3 Polycom 501s) Question 1: I've never had an MPLS network with QoS. Will the call quality Really be as good as ATT assures me it will be with QoS? Assuming we never have more than 5 calls going over the MPS from a branch? Question 2: MPLS are pretty reliable, but last mile connections can be cut, hardware can fail at ATT or whatever. Is this putting too many eggs in one basket? If I lose the headquarters T1s (MPLS or PRI), everyone is down. Would You do it this way? Question 3: (OT) For those who have used an MPLS. How much better throughput for DATA (NOT VoIP) should I see compared to using the Internet? I'm mostly just curious here and realize it is hard to compare, but when I do any type of file transfer between office right now, I use FTP over the Internet and both ends have a T1. Assuming an MPLS on each end, what is your experience when comparing average throughput compared to an Internet transfer. Just a guess of what you've seen. (i.e. Yes, you'll see a big difference, maybe a little better or couldn't really see that big of a difference) Thanks for your input. Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users One of the techs I know built such a setup for a company in australia - it actually became one of the original Asterisk case studies. She had about 200 phones over 70 sites (with between 2 and 5 phones per site) - all over an MPLS network. Her only comments - get decent phones (she used polycoms), use g729. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Asterisk out of the RTP media path
On Aug 11, 2008, at 12:04 PM, SIP wrote: SIP wrote: When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP media path. I understand that this is sort of the idea behind a bridged channel, but is there any way to avoid it? Is there any way to say Connect this number and this number and then get out of the way, or is this a design limitation? No ideas on this one? I've tried everything I can think of and then some and still can't get Asterisk out of the media path. I can do it if I don't originate the call with Asterisk, but only use Asterisk to connect one leg of the call, but if I use Asterisk to connect both legs, no luck. Going about this the wrong way? Asterisk will re-INVITE the media away from itself as long as it doesn't have a reason to need access to the media. For example, if you've enabled call recording, then clearly Asterisk needs access to the media. Other reasons include enabling features controlled via DTMF when the DTMF follows the media path. Nobody can help any further without seeing the details of your configuration. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The problem DIAL with option T,t
On Aug 11, 2008, at 2:03 AM, larry wrote: This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext = 700 parkpos = 701-702 context = parkedcalls [featuremap] atxfer = *2 The most likely cause of why it's not working is that you're not pressing the digits fast enough. The default timeout is 500 ms. So, if you don't press 2 within half a second of pressing *, it won't work. There is an option to extend this timeout - featuredigittimeout, I think. [applicationmap] set(DYNAMIC_FEATURES=tranf) tranf = *2,peer,waitexten(10|m) This is completely unnecessary for configuring call transfer. If you were to configure custom features, though, you would have the Set() command in the dialplan (extensions.conf). -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intermittent T.38 pass through
On August 11, 2008 06:59:23 pm JR Richardson wrote: So my question is this: Can I setup Asterisk to only allow t.38 pass through from these ATA's, without the need to use the #99 in every dial string from the fax machine? Can you use disallow/allow with UDPTL? I'm not sure, I've never played with this before. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 27
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] park calls - cannot hear digits being played
I configured parkcalls and can see on cli Playing digits/7 etc but I cannot hear them in the phone. I have: features.conf. [general] parkext = 700 parkpos = 701-720 extensions.conf: [extensions] include = parkedcalls What did I miss? -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
