[Asterisk-Users] asterisk cluster
Hello, Is it possible to run Asterisk in a cluster? OpenMosix or other cluster software. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime database Problem
Dear Matt, Yes indeed I did I have used cvs to download asterisk and it's addon from CVS. Thx MAG Matthew Boehm wrote: Did you install res_config_mysql.so from asterisk-addons? -Matthew > From: "Mohamed A. Gombolaty" [EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > asterisk-users@lists.digium.com> > Date: Sun, 10 Jul 2005 12:16:51 +0300 > To: Asterisk Users Mailing List - Non-Commercial Discussion > asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] Asterisk Realtime database Problem > > Hi All, > > I am facing a problem with makeing asterisk work realtime with mysql, after > following the tiki steps which are: > > uncommented the lines sipuser and sippeers from extconfig.conf > copied the res_mysql.conf and configured it with the right parameters > checked that mysql is working > added the realtime switch to the extensions.conf > > Now when asterisk is starting I don't see it even to attempt to parse the > res_mysql.conf file so I am assuming that there is something missing what is > it I > don't know. > > -- > Thx > MAG > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] searching for assistance
Hello! I'm quite unsure, if i'm right here with this question... I've a customer with a IVR-based PrePaidSystem (DTMF-control, MySQL) which he wants to port from dialogic/envox on ISDN to a SIP solution. I think this should be solvable by an asterisk-solution - but i have far too low insight. I will need assistance in planning and deciding about feasability and also later in programming, deploying and supporting it. I will prefer someone in germany, best near Hannover (my site) or Chemnitz (customer's site). The job should surely be paid for. best regards, Robert -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.11/45 - Release Date: 09.07.2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnophone installation
i ve tried to find a gnophone dependency libgtksuperwin.so i searched every where in google in wiki pages but i didnt found it at all ,if any one can help in finding it i ll be thankful,and thanks in advance. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video phone settings???
I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phone settings???
Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. Giorgio. Ronald_Wiplinger wrote: I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Video phone settings???
Giorgio Incantalupo wrote: Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. I have already set that. Without that NO video at all at any try. bye Ronald Giorgio. Ronald_Wiplinger wrote: I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
Re: [Asterisk-Users] GXP-2000 MWI
Thanks mate - I had my voicemail context set up wrong cheers - works a treat for me too! ;-) Mark On 7/11/05, Peter Bowyer [EMAIL PROTECTED] wrote: On 10/07/05, Mark Edwards [EMAIL PROTECTED] wrote: anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set up the mailbox in the sip.conf entries but no flashing lights... SIP NOTIFY seems to be being sent out... Yes, works like a charm here. Firmware 1.0.1.9.Peter--Peter BowyerEmail: [EMAIL PROTECTED]Tel: +44 1296 768003VoIP: sip:[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- regards,Mark P. EdwardsFWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On Monday 11 Jul 2005 05:02, Michael Stearne wrote: On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote: Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a ATT Natural voices seem to be pretty good. You can hear samples here: http://www.wizzardsoftware.com/att_desktop.php . The Rich voice I think sounds the best. They are a little better than the Cepstral voices. But the Cepstral voices are vey good also. Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi), the ATT system sounds awful. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sharing variables between contexts
Hello all,
I'm having trouble getting variables to work the way I want them to, let me
begin with a simple explanation of the problem, I'm using something like this
in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1)
exten = 6000,1,SetVar([EMAIL PROTECTED])
exten = 6000,2,Goto(mailexten,s,1)
[mailexten]
exten = s,1,System(/mail.sh ${Recipient})
exten = s,2,Hangup
As an unsuspecting user, I thought this would work - the variable Recipient
should be available in the [mailexten] context, but apparently this is not
the case. I'm using Asterisk 1.0.9, is this a known problem or am I just
expecting the wrong thing?
Sincerely,
Frank
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RE: [Asterisk-Users] Video phone settings???
Hi, -Original Message- disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Have you tried permutations of this ? I have had working setups with everything except h263p. My experience with leadtek phones is they tend to crash when they are talking to any phone model that is not exactly the same (i.e. bugs in decoding perhaps ?) Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAL Event, who picks up?
Dear asterisk-experts, i've got a problem with my Dialplan. The task is to get the SIP-address of the called internal phone, that answered as first. In this example, two phones are ringing: DIAL(SIP/1SIP/2|120|m) But I want to trigger an event, if someone picks up DIAL(SIP/1SIP/2|120|mM(answered)) The Macro function doesn't have any information about the current address of the answered phone. I'm not able to pass ARGs to the macro (i think i've to use the CVS Version), but i don't expect that this is the solution. I don't know what variable to pass to the macro. Does anybody understand my problem? regards Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phone settings???
Hi, It seems that you are using different audio codec (Unknown RTP codec 96 received) Try to use standard audio code. Sometimes telephone use codec with bad rtp code inside. I use alw or ulaw for my test. Marino On 7/11/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Hi, try videosupport=yes in the general section of sip.conf. Maybe it can work. I have already set that. Without that NO video at all at any try. bye Ronald Giorgio. Ronald_Wiplinger wrote: I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004,
[Asterisk-Users] Dial SIP extension
I have 2 Sipuras with the following configuration: The first: SAS Enable: no, NAT Mapping Enable: No, Sip Port 5060, USER ID 1002, Auth ID: 1002, Preferred Codec G711a, Prefered Codec Only: no, DTMF Tx Method, INFO, Enable IP Dialing: no. The second the same with USER ID : 1003 * Name : 1002 Secret : Not set MD5Secret: Not set Context : outgoing Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: 1, 33 Pickupgroup : 1, 33 Mailbox : LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : Yes Callerid : Expire : 243334 Expiry : 900 Insecure : no Nat : Always ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : info LastMsg : 0 ToHost : Addr-IP : x.x.x.x (dont ask) Port 32453 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 1002 Codecs : 0x8 (alaw) Codec Order : (alaw) Status : OK (92 ms) Useragent: Sipura/SPA2000-2.0.10(e) Reg. Contact : sip:[EMAIL PROTECTED]:5061 * Name : 1003 Secret : Not set MD5Secret: Not set Context : outgoing Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: 1, 33 Pickupgroup : 1, 33 Mailbox : LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : Yes Callerid : Expire : 242099 Expiry : 900 Insecure : no Nat : Always ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : info LastMsg : 0 ToHost : Addr-IP : x.x.x.x Port 27495 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 1003 Codecs : 0x8 (alaw) Codec Order : (alaw) Status : OK (90 ms) Useragent: Sipura/SPA2000-2.0.10(e) Reg. Contact : sip:[EMAIL PROTECTED]:5061 Who do I configure * to dial from one to another as an extension in a network? Thanks _ ¿Estás pensando en cambiar de coche? Todas los modelos de serie y extras en MSN Motor. http://motor.msn.es/researchcentre/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls dropped upon 'native bridging' after IAX2 transfer
Hi,I'm facing an issue with my * boxes. Some calls are dropped while bridging after a transfer.I have 2 * boxes, let's call them 'dell' and 'amd', they are connected via IAX2.'dell' has extensions that matches 63XX while 'amd' matches 62XX.Here's an example where call will be dropped. Step 1) Bob (ext. 6202 on 'dell') calls the operator (ext. 6302 on 'amd') Step 2) The operator transfers the call to Brad (ext 6203 on 'dell') 'dell' performs a native bridge and releases IAX channels. On 'amd' calls between 6203 and 6202 are still stated as bridged to IAX2 channels Step 3) Brad transfers the call to Bert (ext. 6303 back on 'amd') 'amd' attempts a native bridge and releases two IAX2 channels, including the one said to be bridged to SIP/6202. Results SIP/6202 is hung up too after few seconds. On 'dell' connexion stays active. Bert has just heard 'allo' from Brad before then '...' (silence) Whithout any 'hung up' notification. All sip peers have the 'canreinvite' option set to 'yes'.I've tried setting "notransfer=yes" in iax.conf, but the result is many, useless, IAX2 channels and longer delays. My goal is to force 'amd' to perform a native bridge, at step 2, between 6203 and 6202. You'll find enclosed to files with my console's output. Enclosed to this mail, you'll find output of my * console's output on both hostsI hope one can help me sort this out.Regards,Luba Vincent # amd BOX # ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup(SIP/6202-d193, IAX) in new stack -- Executing NoOp(SIP/6202-d193, ) in new stack -- Executing GotoIf(SIP/6202-d193, 0?4:7) in new stack -- Goto (from-ip-phones,6302,7) -- Executing SetVar(SIP/6202-d193, NumToDial=6302) in new stack -- Executing Dial(SIP/6202-d193, IAX2/amd:[EMAIL PROTECTED]/6302|300|tTrF) in new stack -- Called amd:[EMAIL PROTECTED]/6302 -- Call accepted by $DELL_IP$ (format gsm) -- Format for call is gsm -- IAX2/$DELL_IP$:4569/1 is ringing -- IAX2/$DELL_IP$:4569/1 answered SIP/6202-d193 amd1-itbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data IAX2/$DELL_IP$:4569/1 ( s1 ) Up Bridged Call SIP/6202-d193 SIP/6202-d193 (from-ip-phones 6302 10 ) Up Dial IAX2/amd:[EMAIL PROTECTED]/6302|300|tTrF 2 active channel(s) ## Step 2 ## The operator transfers the call to Bradd (ext. 6203) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Accepting AUTHENTICATED call from $DELL_IP$, requested format = 2, actual format = 2 -- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569/2, CALLEDID=6203) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569/2, 0?3:7) in new stack -- Goto (from-remote-hosts,6203,7) -- Executing Macro(IAX2/[EMAIL PROTECTED]:4569/2, dialuser|6203|30|tTrF) in new stack -- Executing DBget(IAX2/[EMAIL PROTECTED]:4569/2, temp=FM/6203) in new stack -- DBget: varname=temp, family=FM, key=6203 -- DBget: Value not found in database. -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569/2, SIP/6203|30|tTrF) in new stack -- Called 6203 -- SIP/6203-078d is ringing -- SIP/6203-078d answered IAX2/[EMAIL PROTECTED]:4569/2 amd1-itbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data SIP/6203-078d (from-ip-phones 1 ) Up Bridged Call IAX2/[EMAIL PROTECTED]:4569/2 IAX2/[EMAIL PROTECTED]:4569/2 (macro-dialuser s102 ) Up Dial SIP/6203|30|tTrF IAX2/$DELL_IP$:4569/1 ( s1 ) Up Bridged Call SIP/6202-d193 SIP/6202-d193 (from-ip-phones 6302 10 ) Up Dial IAX2/amd:[EMAIL PROTECTED]/6302|300|tTrF 4 active channel(s) == Spawn extension (from-ip-phones, 6303, 0) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569/2' in macro 'dialuser' == Spawn extension (from-ip-phones, 6303, 0) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569/2' ## Step 3 ## Place a remote IAX2 incoming call from Bob (ext. 6202) to the Bert (ext. 6303) transfered by Brad (ext. 6203) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup(IAX2/[EMAIL PROTECTED]:4569/2, IAX) in new stack -- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569/2, ) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569/2, 0?4:7) in new stack -- Goto (from-ip-phones,6303,7) -- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569/2, NumToDial=6303) in new stack -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569/2, IAX2/amd:[EMAIL PROTECTED]/6303|300|tTrF) in new stack -- Called amd:[EMAIL PROTECTED]/6303 -- Call accepted by $DELL_IP$ (format gsm) -- Format for call is gsm -- IAX2/$DELL_IP$:4569/3 is ringing -- IAX2/$DELL_IP$:4569/3 answered IAX2/[EMAIL
Re: [Asterisk-Users] Sharing variables between contexts
On Mon, 11 Jul 2005, Frank Schoep wrote:
Hello all,
I'm having trouble getting variables to work the way I want them to, let me
begin with a simple explanation of the problem, I'm using something like this
in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1)
exten = 6000,1,SetVar([EMAIL PROTECTED])
exten = 6000,2,Goto(mailexten,s,1)
[mailexten]
exten = s,1,System(/mail.sh ${Recipient})
exten = s,2,Hangup
As an unsuspecting user, I thought this would work - the variable Recipient
should be available in the [mailexten] context, but apparently this is not
the case. I'm using Asterisk 1.0.9, is this a known problem or am I just
expecting the wrong thing?
Try SetGlobalVar()
Armin
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RE: [Asterisk-Users] Video phone settings???
I found the problem was with eyeBeam when I had more than one video codec enabled. Try on eyebeam to only have h263p enabled. Does the video appear in the Echo test? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: Monday, July 11, 2005 12:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video phone settings??? I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] How to force RTP through Asterisk PBX.
Hi. I would like to force RTP traffic for SIP to go through PBX. Is it possible to somehow force it in configuration? Is there also possible for IAX? Regards Marcin Okraszewski ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to download chan_sip2
I believe you can find it here: http://bugs.digium.com/bug_view_page.php?bug_id=759 S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad Sent: Sunday, July 10, 2005 10:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to download chan_sip2 hello http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+chan_sip2 where can i download chan_sip2.c thanks Kamran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to force RTP through Asterisk PBX.
For SIP in sip.conf use the option canreinvite=no for a specifed device. For IAX2 I think you have to use notransfer=yes Ivan I would like to force RTP traffic for SIP to go through PBX. Is it possible to somehow force it in configuration? Is there also possible for IAX? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk as media proxy
hi, i need instructions on how to configure asterisk as a media server. i need your help. thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with a new box of [EMAIL PROTECTED] 1.3
Hi, I got having problem in my asterisk when i call i always see this and degrade the voice quality of the call.how can i resolve thisplease help Jul 11 19:37:39 WARNING[74771]: samples/codec_g729.c:217 g729tolin_framein: Received a G.729 frame that was 2 bytes from RTP regards, jollyr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2.6.13 Kernels
First, thanks to Kevin for the quick response to the 'minor' problem that zaptel had with 2.6.13 kernels. Interestingly in the kernel config you can now change the timer frequency. According to the help messages 100 HZ is a typical choice for servers, SMP and NUMA systems with lots of processors that may show reduced performance if too many timer interrupts are occurring. 250 HZ is a good compromise choice allowing server performance while also showing good interactive responsiveness even on SMP and NUMA systems. 1000 HZ is the preferred choice for desktop systems and other systems requiring fast interactive responses to events. It will be interesting to see how this works with *. Anyone else trying 2.6.13? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On Mon, 2005-07-11 at 09:19 +0100, Bob Goddard wrote: Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi), the ATT system sounds awful. Yes it does sound considerably better, but what do I know I have a hearing loss. Anyway, have you managed to integrate this with asterisk successfully? If so how? use the generated output (wav presumably) and stream it via an AGI? Or does it have more direct asterisk connectivity? I had written a script that would snarf sitepal TTS data, rip the SWF to get the resulting mp3, transcode that to something more appropriate, and cache it for future requests of the same data. I called this script via a macro so it was trivial to specify a voice and text to speak in a dialplan entry. The whole process was quite trivial but not as system friendly as I like (too many fork and execs for my taste). Plus becuase the TTS engine runs on their systems there is noticable lag in actually getting the data - part of which appears to be their system is somewhat slow in generating the audio data, but in this instance free. I could of course do something similar with Rhetorical, even if its running on my own machine, but I just dont like doing it that way, it works, but seems wrong due to many wasted cpu cycles. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi), the ATT system sounds awful. You're 100% correct! My mistake, I was thinking of rhetorical when I said ATT. I'm not familiar with ATT at all - my bad! Thanks for correcting this and reminding me of rhetorical. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sharing variables between contexts
I am having the same thing on my extensions.conf and it works fine. I am
using Asterisk 1.0.7
On Mon, 11 Jul 2005 12:04:59 +0200 (CEST), Armin Schindler wrote
On Mon, 11 Jul 2005, Frank Schoep wrote:
Hello all,
I'm having trouble getting variables to work the way I want them to, let
me
begin with a simple explanation of the problem, I'm using something like
this
in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1)
exten = 6000,1,SetVar([EMAIL PROTECTED])
exten = 6000,2,Goto(mailexten,s,1)
[mailexten]
exten = s,1,System(/mail.sh ${Recipient})
exten = s,2,Hangup
As an unsuspecting user, I thought this would work - the variable
Recipient
should be available in the [mailexten] context, but apparently this is
not
the case. I'm using Asterisk 1.0.9, is this a known problem or am I just
expecting the wrong thing?
Try SetGlobalVar()
Armin
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Re: [Asterisk-Users] Video phone settings???
Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. Regards. On 7/11/05, Storm D. J. Petersen [EMAIL PROTECTED] wrote: I found the problem was with eyeBeam when I had more than one video codec enabled. Try on eyebeam to only have h263p enabled. Does the video appear in the Echo test? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: Monday, July 11, 2005 12:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video phone settings??? I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on
Re: [Asterisk-Users] Voicemail = SMS
On Thu, 2005-06-30 at 23:34 -0700, snacktime wrote: The manager action MailboxCount gives the number of old and new messages in a mailbox. You would have to call the manager via an agi but it would give you the info you want. The count is given as an argument to the voicemailnotify program. I just have this in voicemail.conf... externnotify=/etc/asterisk/voicemailnotify.sh ... and a trivial shell script which does something like this... #!/bin/sh logger voicemail $@ MOBILE_NR=07976xxx if [ $3 = 0 ]; then smsq --dcs=0xc0 $MOBILE_NR else smsq --dcs=0xc8 $MOBILE_NR fi I've hardcoded the phone number but the mailbox identity is in $2 when the script is invoked, so it wouldn't be hard to look it up from a table. -- dwmw2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] searching for assistance
Try over on the Asterisk-biz forum Robert Schulz wrote: Hello! I'm quite unsure, if i'm right here with this question... I've a customer with a IVR-based PrePaidSystem (DTMF-control, MySQL) which he wants to port from dialogic/envox on ISDN to a SIP solution. I think this should be solvable by an asterisk-solution - but i have far too low insight. I will need assistance in planning and deciding about feasability and also later in programming, deploying and supporting it. I will prefer someone in germany, best near Hannover (my site) or Chemnitz (customer's site). The job should surely be paid for. best regards, Robert -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto get streaming mp3 at an extension?
Easy!! Add another line to your musiconhold.conf file like this stream = /var/lib/asterisk/stream,http://sourcepfstream.com:8001/ Then add an externsion number to extensions.conf that uses the stream variable to play the hold music. There's quite a bit about this in the wiki. Mark [EMAIL PROTECTED] wrote: I would simply like to dial an extension and get an individual Live MP3 stream but am unsure of how to do this. I'd like it to be different from my music on hold (not the same source) This trick works for music on hold: in musiconhold.conf ;default = mp3:/var/lib/asterisk/live,http://sourceofstream.com:8001/ I still wish to use local files for music on hold but want to dial an extension to listen to a live stream. Any ideas? Thanks! Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to voicemail...
Also, the most annoying is when I try to place an outside call...
At that point, if the call is to a regular phone number, I'll get a message
asking me to enter my password... followed by pound...
Any idea, how I can resolve this issue? I feel like i'm very close to getting
this thing working, but I'm pretty frustrated because I don't know what to edit
or change at this point...
Below is my configurational files..
+
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding nat=1 to each peer definition to
; solve translation problems.
[general]
externip=69.165.XXX.XXX
localnet=192.168.1.105/255.255.255.0
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
register =
[EMAIL PROTECTED]:MYSECRETPASS:[EMAIL PROTECTED]/200
[sip.broadvoice.com]
type=peer
;Enter your closest proxy server
host=proxy.mia.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=3055741428
secret=MYSECRETPASS
context=from-broadvoice
;Disable canreinvite if you are behind a NAT
canreinvite=no
;Don't try to authenticate on incoming calls
insecure=very
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Systems Inc
; dialparties.agi (»www.sprackett.com/asterisk/)
; Asterisk::AGI (»asterisk.gnuinter.net/)
; gsm (»www.ibiblio.org/pub/Linux/utils/compre..)
; loligo sounds (»www.loligo.com/asterisk/sounds/)
; mpg123 (»voip-info.org/wiki-Asterisk+config+mus..)
; include extension contexts generated from AMP
#include extensions_additional.conf
; Customizations to this dialplan should be made in extensions_custom.conf
; See extensions_custom.conf.sample for an example
#include extensions_custom.conf
[from-trunk] ; just an alias since VoIP shouldn't be called PSTN
include = from-pstn
[from-pstn]
include = from-pstn-custom ; create this context in extensions_custom.conf to
include customizations
include = ext-did
include = from-pstn-timecheck ; this has to be included otherwise it overrides
ext-did
[from-pstn-timecheck]
exten = .,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a
DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1
exten = s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1
exten = s,4,Goto(from-pstn-afthours,s,1)
[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2) ; if
fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1)
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING starts with EXT, then
assume its an extension
exten = s,7,Wait(3) ;wait 3 more second to make sure this isn't a fax before
dialing someone
exten = s,8,Goto(ext-local,${INCOMING:4},1)
exten = s,9,GotoIf($[${intype} = GRP]?10:12) ; If INCOMING starts with GRP,
then assume its a ring group
exten = s,10,Wait(3)
exten = s,11,Goto(ext-group,${INCOMING:4},1)
exten = s,12,GotoIf($[${intype} = QUE]?13:15)
exten = s,13,Wait(3)
exten = s,14,Goto(ext-queues,${INCOMING:4},1)
exten = s,15,Goto(${INCOMING},s,1) ; not EXT or GR1 - it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup
[from-pstn-reghours-nofax]
exten = s,1,SetVar(intype=${INCOMING})
exten = s,2,Cut(intype=intype,-,1)
exten = s,3,GotoIf($[${intype} = EXT]?4:5) ; If INCOMING starts with EXT, then
assume its an extension
exten = s,4,Goto(ext-local,${INCOMING:4},1)
exten = s,5,GotoIf($[${intype} = GRP]?6:7) ; If INCOMING starts with GRP, then
assume its a ring group
exten = s,6,Goto(ext-group,${INCOMING:4},1)
exten = s,7,GotoIf($[${intype} = QUE]?8:11) ;queue
exten = s,8,Answer ; answer call before queue
exten = s,9,Wait(1)
exten =
Re: [Asterisk-Users] 2.6.13 Kernels
Dave Cotton wrote: First, thanks to Kevin for the quick response to the 'minor' problem that zaptel had with 2.6.13 kernels. You are welcome :-) 100 HZ is a typical choice for servers, SMP and NUMA systems with lots of processors that may show reduced performance if too many timer interrupts are occurring. It's always bugged me that my servers have to run with 1000Hz timer frequency just because someone decided that was better for interactive performance, so I consider this a big improvement. 250 HZ is a good compromise choice allowing server performance while also showing good interactive responsiveness even on SMP and NUMA systems. And this appears to be the default, as well. This means that USE_RTC in ztdummy is going to be mandatory for 2.6.13+. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sharing variables between contexts
Try prepending two _'s like this.
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1
It allows the variable to be exported.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Frank Schoep
Sent: Monday, July 11, 2005 4:40 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Sharing variables between contexts
Hello all,
I'm having trouble getting variables to work the way I want
them to, let me begin with a simple explanation of the
problem, I'm using something like this in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1)
exten = 6000,1,SetVar([EMAIL PROTECTED])
exten = 6000,2,Goto(mailexten,s,1)
[mailexten]
exten = s,1,System(/mail.sh ${Recipient}) exten = s,2,Hangup
As an unsuspecting user, I thought this would work - the
variable Recipient should be available in the [mailexten]
context, but apparently this is not the case. I'm using
Asterisk 1.0.9, is this a known problem or am I just
expecting the wrong thing?
Sincerely,
Frank
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Re: [Asterisk-Users] iax.cc opinion request
[EMAIL PROTECTED] wrote: On 7/10/2005, trixter wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Well - here we have a quandary. Opinion? Bad. (But so good) No outages that I can place on Sixtel - 24/7 rock solid - think a router hiccupped once for a couple of hours, but it wasn't theirs. Packet loss - again - as good or better that cell phones. Can't fault them (or him) there. UK termination (DID?) - can't say - thought they (or him) were US only. Tech support? Hahahahahahahahahahaha Ouch - my sides hurt! Took a month to get a DID. This pretty much sums it up for me as well. Except that it took two months for my DID to become active. On the other hand, I've had zero downtime and my 800 number was active within a day. I'm not noticing any problems with call quality either. They claimed in an email from early June to have instituted a new support responsiveness guarantee, but like I said earlier, since the DID went active I've had zero (!) problems. The server that you would be connecting to is iax2.sixtel.net, so run a few tracroutes from your site. Best, Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime database Problem
Mohamed A. Gombolaty wrote: Dear Matt, Yes indeed I did I have used cvs to download asterisk and it's addon from CVS. If you followed these instructions then it should be working: cd /usr/src/asterisk make; make install cd /usr/src/asterisk-addons make; make install cp configs/res_mysql.conf /etc/asterisk/ Have you enabled debug log in logger.conf? That might be helpful. Do you have autoload turned on in modules.conf? Type this from asterisk CLI: realtime mysql status If that command is unrecognized, then you have not installed the module. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phone settings???
apenon apenon wrote: Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. How to make the echo test? bye Ronald Wiplinger Regards. On 7/11/05, Storm D. J. Petersen [EMAIL PROTECTED] wrote: I found the problem was with eyeBeam when I had more than one video codec enabled. Try on eyebeam to only have h263p enabled. Does the video appear in the Echo test? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: Monday, July 11, 2005 12:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video phone settings??? I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102
Re: [Asterisk-Users] Valgrind effects
Benjamin Lawetz wrote: I have a couple of bugs I'm trying to debug compiling asterisk with valgrind. But of course when compiled like that the bugs don't occur. What are the exact effects of Valgrind? Would there be a hit on performance running asterisk compiled with valgrind ? 'make valgrind' turns off all compiler optimizations. Yes, there will be a performance hit, if your system is heavily loaded you will be able to handle fewer calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2.6.13 Kernels
On Mon, 2005-07-11 at 08:31 -0500, Kevin P. Fleming wrote: 100 HZ is a typical choice for servers, SMP and NUMA systems with lots of processors that may show reduced performance if too many timer interrupts are occurring. It's always bugged me that my servers have to run with 1000Hz timer frequency just because someone decided that was better for interactive performance, so I consider this a big improvement. 250 HZ is a good compromise choice allowing server performance while also showing good interactive responsiveness even on SMP and NUMA systems. And this appears to be the default, as well. This means that USE_RTC in ztdummy is going to be mandatory for 2.6.13+. I'm now running at 100. Just seems odd to have cat /proc/interrupts saying that timer is 1/10th of wcfxo/wctdm as they've always been roughly the same count. I just hope it sorts out my lockups on the tdm or fxo cards. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.cc opinion request
On Mon, 2005-07-11 at 09:40 -0400, David Mallwitz wrote: This pretty much sums it up for me as well. Except that it took two months for my DID to become active. On the other hand, I've had zero downtime and my 800 number was active within a day. I'm not noticing any problems with call quality either. They claimed in an email from early June to have instituted a new support responsiveness guarantee, but like I said earlier, since the DID went active I've had zero (!) problems. The server that you would be connecting to is iax2.sixtel.net, so run a few tracroutes from your site. Thanks, (and to the others that commented as well). I plan on using them for outbound only, so the issues about DIDs not working does not mean much to me, although the underlying problems do give me cause for concern. I would like to think that when something does break (it will eventually no matter what format of transmission is used (pstn or voip), no matter which company) someone will be there to fix it, however the fact that it usually works seems to be better than some providers. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as Gateway
Hello to all
I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with
CAPI) to connect to a Siemens PBX, but I still cant forward calls to the
Siemens PBX (neither receive them from the PBX).
Here s the result in the asterisk console when I try to dial the 116 PBX
phone:
-- Executing Dial(SIP/193.136.2.205:5060-fd1f,
CAPI/12345678:b116|90) in new stack
-- data = 12345678:b116
-- capi request omsn = 12345678
== found capi with omsn = 12345678
== CAPI Call CAPI[contr1/12345678]/11 with B3-- Called 12345678:b116
-- CONNECT_CONF ID=001 #0x0012 LEN=0014
Controller/PLCI/NCCI= 0x301
Info= 0x0
-- CONNECT_CONF ID=001 #0x0012 LEN=0014
Controller/PLCI/NCCI= 0x301
Info= 0x0
== received CONNECT_CONF PLCI = 0x301 INFO = 0
-- DISCONNECT_IND ID=001 #0x001b LEN=0014
Controller/PLCI/NCCI= 0x301
Reason = 0x3302
== DISCONNECT_IND PLCI=0x301 REASON=0x3302
-- CAPI Hangingup
== No one is available to answer at this time
this is my CAPI.CONF
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
-this is my EXTENSIONS.CONF
[from-sip]
exten = _XXX,1,Dial,CAPI/12345678:b${EXTEN}|90
Does someone have an ideia of what is missing?
The Siemens PBX should forward the call to its 116 extension... but
there's no way I can debug it...
Joao Pereira
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[Asterisk-Users] Trunk number (SMDI)
Hello! I'm integrating an Asterisk-based voicemail system with an old switch, and I want the call history from SMDI. My understanding is that the terminal number in the SMDI message matches the channel's trunk number. From within an Asterisk app, how do I get the trunk number? Thanks! Nate ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] + Broadvoice = Almost working installation...
Is your server behind a NAT? If so, make sure that you have configured
/etc/asterisk/sip_nat.conf with your proper settings (change the
localnet and externip settings to match your setup):
nat=yes
externip=xxx.xxx.xxx.xxx
localnet=10.0.0.0/255.255.255.0
sip_nat.conf may only affect your extensions, not Broadvoice, but you
also need to make sure that your NAT is forwarding the relevant ports
through to your server for SIP (5060, 1-2, if I remember
correctly)
Finally, look at the geekgazette article for the proper way to set up
your Broadvoice Trunk in AMP:
http://geekgazette.com/index.php?
option=com_contenttask=viewid=20Itemid=31
Also, it would help to double-check that your system is indeed
registering with Broadvoice by running the command sip show registry
at the command asterisk prompt.
Tom
On Jul 11, 2005, at 9:18 AM, [EMAIL PROTECTED] wrote:
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed
up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions,
but whenever
anyone tries to call my broadvoice account, they are greeted by no
ring or
anything, but rather simply a direct to voicemail...
Also, the most annoying is when I try to place an outside call...
At that point, if the call is to a regular phone number, I'll get a
message
asking me to enter my password... followed by pound...
Any idea, how I can resolve this issue? I feel like i'm very close to
getting
this thing working, but I'm pretty frustrated because I don't know
what to edit
or change at this point...
Below is my configurational files..
+
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding nat=1 to each peer definition to
; solve translation problems.
[general]
externip=69.165.XXX.XXX
localnet=192.168.1.105/255.255.255.0
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
register =
[EMAIL PROTECTED]:MYSECRETPASS:
[EMAIL PROTECTED]/200
[sip.broadvoice.com]
type=peer
;Enter your closest proxy server
host=proxy.mia.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=3055741428
secret=MYSECRETPASS
context=from-broadvoice
;Disable canreinvite if you are behind a NAT
canreinvite=no
;Don't try to authenticate on incoming calls
insecure=very
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS+
EXTENSIONS+ EXTENSIONS+ EXTENSIONS+ EXTENSIONS+
EXTENSIONS
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Systems Inc
; dialparties.agi (»www.sprackett.com/asterisk/)
; Asterisk::AGI (»asterisk.gnuinter.net/)
; gsm (»www.ibiblio.org/pub/Linux/utils/compre..)
; loligo sounds (»www.loligo.com/asterisk/sounds/)
; mpg123 (»voip-info.org/wiki-Asterisk+config+mus..)
; include extension contexts generated from AMP
#include extensions_additional.conf
; Customizations to this dialplan should be made in
extensions_custom.conf
; See extensions_custom.conf.sample for an example
#include extensions_custom.conf
[from-trunk] ; just an alias since VoIP shouldn't be called PSTN
include = from-pstn
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to
include customizations
include = ext-did
include = from-pstn-timecheck ; this has to be included otherwise it
overrides
ext-did
[from-pstn-timecheck]
exten = .,1,Goto(s,1) ; catch-all matching for calls that have DID
info (if a
DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1
exten =
s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1
exten = s,4,Goto(from-pstn-afthours,s,1)
[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} =
disabled]?from-pstn-reghours-nofax,s,1:2) ; if
fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1)
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING starts with
EXT, then
assume its an extension
exten = s,7,Wait(3) ;wait 3 more second to make sure this isn't a fax
before
dialing someone
exten = s,8,Goto(ext-local,${INCOMING:4},1)
exten =
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
On Thursday 23 June 2005 2:57am, Patrick Lidstone wrote: I have a second-hand 7960 which I am attempting to upgrade to use a SIP image. The phone currently has a firmware release which doesn't seem to be listed in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests the firmware image listed in OX79XX.txt correctly, displaying Upgrading Software on the screen. It then continues to re-request the same image from the tftp server at 10s intervals indefinitely. What am I doing wrong? Patrick, did you ever find a solution for this? I am currently experiencing the same problem and am not having any success in locating a solution. -- Marc H. Fishman OuttaSite Resources [EMAIL PROTECTED] If you woke up breathing, congratulations! You have another chance! PGP KeyID: 6C8E212E75CDBD79 PGP Key Fingerprint: E620 1F11 D3AC 6FEC 4CC5 8CA6 6C8E 212E 75CD BD79 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sharing variables between contexts
On Monday 11 July 2005 14:33, jurczak wrote:
I am having the same thing on my extensions.conf and it works fine. I am
using Asterisk 1.0.7
Is it possible that using queues causes problems with regard to handling
variables? It seems that variable handling between contexts is broken after
an incoming call has gone through a queue. Even handling variables within a
context seemed to go foobar after a queue, at least it did when I tested it.
I'll try investigating what causes the problem, but I've currently implemented
a workaround using the ${CALLERIDNAME} channel variable.
Sincerely,
Frank
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[Asterisk-Users] Vikrant Mathur lead developer for the open source OSP Toolkit to speak at Cluecon.
Vikrant Mathur is the lead developer for the open source OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer developing solutions for secure peer to peer routing, access control and accounting of VoIP traffic on the Internet. If you haven't registered yet please do so ASAP so we can make sure to reserve you a room! Thanks, Brian West http://www.cluecon.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Gateway
On Mon, 11 Jul 2005, Joao Pereira wrote: Hello to all I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with CAPI) to connect to a Siemens PBX, but I still cant forward calls to the Siemens PBX (neither receive them from the PBX). Here s the result in the asterisk console when I try to dial the 116 PBX phone: -- Executing Dial(SIP/193.136.2.205:5060-fd1f, CAPI/12345678:b116|90) in new stack -- data = 12345678:b116 -- capi request omsn = 12345678 == found capi with omsn = 12345678 == CAPI Call CAPI[contr1/12345678]/11 with B3-- Called 12345678:b116 -- CONNECT_CONF ID=001 #0x0012 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 -- CONNECT_CONF ID=001 #0x0012 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 -- DISCONNECT_IND ID=001 #0x001b LEN=0014 Controller/PLCI/NCCI= 0x301 Reason = 0x3302 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup == No one is available to answer at this time Does someone have an ideia of what is missing? The Siemens PBX should forward the call to its 116 extension... but there's no way I can debug it... I assume you use chan_capi-0.3.5 !? Some messages are missing in the debug, please try chan_capi-0.5.3 from sourceforge. (But note, the capi.conf and dial syntax has changed in that version). Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk @ Home Voicemail
OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B interface to POTS lines. Everything seems to be working just fine, but I have some questions on how to access voicemail options. I can leave a message for an extension, but when I try to retrieve it by using *97 it asks for the password and even though I type in the same password I gave the extension for vmail, it tells me the password is incorrect. I would also like some ideas on how to personalize the greetings for vmail as well. I am a bit of a newbie, so go easy... :-) Thanks, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-Dev] Vikrant Mathur lead developer for the open sourceOSP Toolkit to speak at Cluecon.
Vikrant Mathur is the lead developer for the open source OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer developing solutions for secure peer to peer routing, access control and accounting of VoIP traffic on the Internet. If you haven't registered yet please do so ASAP so we can make sure to reserve you a room! Thanks, Brian West http://www.cluecon.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS Handler in Asterisk
On Sun, 2005-07-10 at 18:43 +0200, Michiel van Baak wrote: [snip] This won't answer your question, sorry. How are you sending SMS ? I'm in NL too, and can't seem to find a way to send SMS with asterisk. The only way I found was some service on the internet that sells SMS credits for asterisk users but it would be nice to know how you are doing it. If the gsm owner has activated the email2sms service than you can send an sms message to gsm number@gin.nl. This will cause the gsm owner to be charged so many have it turned off. Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP PHONE
Try www.SIPphone.com or www.terracall.com Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ellafi FituriSent: Tuesday, July 05, 2005 2:07 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] SIP PHONE Hi All, I just got a SIP phone and I would like to know where I could find service? Please helpThank you very much for your help Yahoo! SportsRekindle the Rivalries. Sign up for Fantasy Football NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error related to the native formats
Hello friends, i make a call through queue to the agent when agent lifts the call it gives one side voice and i get this message in the debug chan_sip.c:1880 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) in my sip.conf iam allowing only ulaw can any body suggest where i may be missing with regards RK __ How much free photo storage do you get? Store your friends 'n family snaps for FREE with Yahoo! Photos http://in.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No sound when dialing out over SIP Proxy
Hi,
i have trouble to dial out over my sip-provider gmx.
I can register with my provider over port 5060 and also dial out.
It rings at the remote phone but when the call is answered there is no
sound / voice to hear.
This is the part from my sip.conf and extensions.conf:
register = 12345:[EMAIL PROTECTED]
[gmx-out]
type=peer
secret=12345
username=12345
host=sip.gmx.net
fromuser=12345
fromdomain=sip.gmx.net
disallow=all
allow=alaw
allow=ulaw
allow=g729
exten = _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Thanks,
kib
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Re: [Asterisk-Users] Problems with a new box of [EMAIL PROTECTED] 1.3
Thanks for all answers. I begin telling you that I'm new to the asterisk world, so I started with [EMAIL PROTECTED] because of its easy installation and setup. I'm doing my tests at home. I have a local network 192.168.1.0 class C (255.255.255.0), my asterisk box has IP 192.168.1.42 and my computer has IP 192.168.1.51 (my DHCP server releases IPs from 51 to 100). As the manual says I downloaded X-Ten SIP soft phone and intalled it. Then I installed [EMAIL PROTECTED] and chaged the IP using netconfig to 192.168.1.42 then restarted. After the restart I entered by web to setup asterisk. As the manual says I've configured externsion 200 with password abc123 (I've deleted the previous extension to configure it again) Then in X-Ten I configured this way in the default SIP: Enabled: Yes Username: 200 Authorization user: 200 Password: abc123 SIP Proxy: 192.168.1.42 Out bound proxy: 192.168.1.42 I didn't touch anything else. As I mentioned Zaptel service doesn't works but I don't think this is the problem because thanks to Steve I know this service is for meetme. Thanks guys for your anwers. Fabrizzio Valencia - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, July 10, 2005 10:53 PM Subject: Re: [Asterisk-Users] Problems with a new box of [EMAIL PROTECTED] 1.3 Hi Welcome to Asterisk On Sun, Jul 10, 2005 at 09:45:19PM -0500, Fabrizzio Valencia wrote: Hello, I've recently installed [EMAIL PROTECTED], i'm following step by step the new user guide but I cannot get my X-Lite SIP phone see my [EMAIL PROTECTED] proxy... I've installed in a viertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Here's another guide to follow: http://www.catb.org/~esr/faqs/smart-questions.html Specifically: http://www.catb.org/~esr/faqs/smart-questions.html#beprecise Please describe what actually happens, and not what doesn't. Configuration files snippets may help (mind the passwords!) . Domps from the asterisk CLI may help as well: script logfile asterisk -vvvr [observe, run, whatever] [press ctrl-c] [press ctrl-d] A dump of that session is now in 'logfile'. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connect 30 phone lines to asterisk how to
On Wed, 2005-07-06 at 16:27 -0300, Angel Diaz wrote: Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel. That all depends on what type of lines you have. If you have an E1 digital trunk then you use a TE100P card in E1 mode. A TE405P will handle 4 E1 for a total of 120 lines. If you only have analog trunks then you need to buy a TE110P and a channelbank for every 24 trunks you want to connect. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and seimens hipath 3750
Hello I am planning to build a small PBX using TDM22B. We have a Siemens Hipath 3750 in operation already. When I manage to complete my PBX using TDM22B I would ofcourse like to be able to connect my Asterisks PBX with the Siemens Hipath 3750 PBX. Will there be any issues regarding my plan ? Or is there any other issues that I need to take into account vis-a-vis Siemens PBX. I have never done all this before so I would appreciate details. Thanks in advance Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
Marc Fishman ha scritto: the firmware image listed in OX79XX.txt correctly, displaying Upgrading Software on the screen. It then continues to re-request the same image from the tftp server at 10s intervals indefinitely. What am I doing wrong? You need to upgrade to a older version first. version 5 or 6 before upgrading it to version 7. Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel configuration for Argentina
I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is the following: /etc/zaptel.conf loadzone=us defaultzone=us # The first Zaptel card has the FXS port. fxoks=1 fxsks=2-8 /etc/asterisk/zapata.conf [channels] rxgain=0.0 txgain=0.0 musiconhold=default busydetect=yes busycount=5 callprogress=yes echocancel=yes ; Internal FAX machine (FXS #1) signalling=fxo_ks language=en immediate=no callwaiting=yes context=pstn-outbound-fax channel = 1 ; Fax phone line (FXO #8) signalling=fxs_ks language=en group=2 callerid=asreceived context=pstn-inbound-fax channel = 8 ; Voice phone lines (FXO #2, #3, #4, #5, #6, #7) signalling=fxs_ks language=en callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 group=1 useincomingcalleridonzaptransfer=yes callerid=asreceived context=pstn-inbound-voice channel = 2-7 Thanks -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DELL 2800 : PCI Parity error
We experienced the same problem on a Dell 2850 server. Our other asterisk admin went a different route and inquired with Dell. They told him this was completely normal and not to worry about it. I'm still skeptical. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Syed Akbar Sent: Sunday, July 10, 2005 7:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DELL 2800 : PCI Parity error I had the same problem with a Dell PowerEdge 800 server and a TDM400 card. I talked to Digium and they suggested a workaround by adding a NMI flag reset in the Linux boot file. This only prevents a system lockup. The system worked fine even with the blinking orange light and the dazed and confused comment from the modprobe command. I have heard that the new firmware on the TDM400P card has fixed this problem, but have not experienced that first hand. In the same machine I am using a new TE110P with no problems at all. Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Sunday, July 10, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DELL 2800 : PCI Parity error I too had this problem, on a 2850, as well as the occasional missed IRQ. I went through all the usual zaptel tuning stuff Disabled fb, disabled ht, disabled acpi (left io-apic enabled), then moved irq affinity of zaptel card to second CPU so all interrupts from zaptel are on their own. My systems now run close to 100% in zttest, never miss an irq and don't seem to generate PCI parity errors any more. I don't know if I've fixed it, but you should really go through the whole process anyway. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == list wrote: Still not resolved On Wed, 2005-06-08 at 01:16, David John Walsh wrote: Frank Did you ever resolve this? If so what was the issue? On 03/05/05, list [EMAIL PROTECTED] wrote: Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no alarm. Zapata Telephony Interface Registered on major 196 Registered Tormenta2 PCI Controller version: 24 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Freshmaker version: 71 Freshmaker passed register test Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips Module 0: Installed -- AUTO FXS/DPO Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) TE110P: Span configured for CCS/HDB3/CRC4 Calling startup (flags is 4099) Registered tone zone 8 (Norway) ramchip problem is false, without the card all ok, ramtests on machine as well. lsmod shows wcusb driver on zaptel, I dont need that, can I remove it? is that a problem or not? # lsmod Module Size Used byNot tainted usbserial 23964 0 (autoclean) (unused) lp 9156 0 (autoclean) parport38848 0 (autoclean) [lp] autofs416984 0 (autoclean) (unused) wcusb 19552 0 (unused) wctdm 41088 0 (unused) wcte11xp 22048 0 (unused) zaptel182080 4 [wcusb wctdm wcte11xp] e1000 77884 1 (autoclean) floppy 57552 0 (autoclean) sg 37388 0 (autoclean) microcode 6912 0 (autoclean) ide-cd 34016 0 (autoclean)
FW: [Asterisk-Users] Retrieving dtmf, passing to shell and getting the result
Title: FW: [Asterisk-Users] Retrieving dtmf, passing to shell and getting the result
John thanks for the help. When I change my plan to this and then dial 2 it gives me a busy signal. When troubleshooting I added an exten = 2,1 Ringing (just as a check) it rang and went straight to busy. On the console I got:
--Executing Ringing(SIP/-00816800, ) in new stack
==Spawn extension(default, 2, 2) exited non-zero on SIP/-00816800
Any ideas?
Jane
Jane,
try this
exten = 2,2,read (firstnumber,enter-first,5)
exten = 2,3,read (secondnumber,enter-second,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber}
${secondnumber})
I believe it is the syntax that is holding you back.
John M
Original Post
I have my asterisk server up and running on OS X and now need to add the
capability to play a sound file asking for a 5 digit number, play another
message asking for a 2 digit number, pass these variables to a shell
script, and get the result. I have tried a number of different scenarios
but they are not working. I have read through the wiki, past posts, and
numerous websites.
The sound files are enter-first enter-second
The shell script is CheckNumbers.sh
exten = 2,2,get_data (enter-first,1,5)
exten = 2,3,get_data (enter-second,1,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber,
secondnumber)
I really appreciate your help!
Jane
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[Asterisk-Users] chan_cornet status
Hi, what is the status of chan_cornet? Does someone here use it in production? I can't find enough info about it. Some URLs will be great. Thank you, -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile Error chan_sccp-20050705 on asterisk 1.0.9 (tarball)
Hello! I tried to compile chan_sccp-20050705 but I receive the following errors: linux:/home/share/chan_sccp-20050705 # make install sh ./create_config.sh /usr/include Checking Asterisk version... * no 'struct ast_channel_tech', using old pvt * no 'struct ast_callerid' * no 'AST_CONTROL_HOLD' * no 'ast_config_load' * no 'ast_copy_string' config.h complete. Now compiling sccp_actions.c 853 lines sccp_actions.c: In function `sccp_handle_unregister': sccp_actions.c:124: parse error before `*' sccp_actions.c:125: `r1' undeclared (first use in this function) sccp_actions.c:125: (Each undeclared identifier is reported only once sccp_actions.c:125: for each function it appears in.) sccp_actions.c: In function `sccp_handle_stimulus': sccp_actions.c:300: parse error before `stimulus' sccp_actions.c:303: `stimulus' undeclared (first use in this function) sccp_actions.c:303: `line' undeclared (first use in this function) sccp_actions.c:308: warning: unreachable code at beginning of switch statement sccp_actions.c: In function `sccp_handle_keypad_button': sccp_actions.c:630: parse error before `int' sccp_actions.c:634: `event' undeclared (first use in this function) sccp_actions.c:638: `resp' undeclared (first use in this function) sccp_actions.c:654: `len' undeclared (first use in this function) sccp_actions.c: In function `sccp_handle_soft_key_event': sccp_actions.c:675: parse error before `*' sccp_actions.c:681: `event' undeclared (first use in this function) sccp_actions.c:682: `line' undeclared (first use in this function) sccp_actions.c:682: `callid' undeclared (first use in this function) sccp_actions.c:688: `l' undeclared (first use in this function) sccp_actions.c:691: `c' undeclared (first use in this function) sccp_actions.c:695: warning: unreachable code at beginning of switch statement sccp_actions.c: In function `sccp_handle_open_receive_channel_ack': sccp_actions.c:747: parse error before `struct' sccp_actions.c:751: warning: built-in function `sin' used without declaration sccp_actions.c:751: request for member `sin_family' in something not a structure or union sccp_actions.c:752: request for member `sin_addr' in something not a structure or union sccp_actions.c:752: request for member `sin_addr' in something not a structure or union sccp_actions.c:753: request for member `sin_port' in something not a structure or union sccp_actions.c:758: `iabuf' undeclared (first use in this function) sccp_actions.c:758: request for member `sin_addr' in something not a structure or union sccp_actions.c:759: request for member `sin_port' in something not a structure or union sccp_actions.c:759: request for member `sin_port' in something not a structure or union sccp_actions.c:759: request for member `sin_port' in something not a structure or union sccp_actions.c:759: request for member `sin_port' in something not a structure or union sccp_actions.c:762: `c' undeclared (first use in this function) sccp_actions.c:764: warning: passing arg 2 of `ast_rtp_set_peer' from incompatible pointer type sccp_actions.c:765: request for member `sin_addr' in something not a structure or union sccp_actions.c:765: request for member `sin_port' in something not a structure or union sccp_actions.c:765: request for member `sin_port' in something not a structure or union sccp_actions.c:765: request for member `sin_port' in something not a structure or union sccp_actions.c:765: request for member `sin_port' in something not a structure or union sccp_actions.c:759: warning: `__v' might be used uninitialized in this function sccp_actions.c:765: warning: `__v' might be used uninitialized in this function sccp_actions.c: In function `sccp_handle_forward_stat_req': sccp_actions.c:838: parse error before `*' sccp_actions.c:839: `r1' undeclared (first use in this function) make: *** [.tmp/sccp_actions.o] Error 1 linux:/home/share/chan_sccp-20050705 What is the problem? -- ciao Holger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Confernce Volume Issues
Hi, I'm hoping someone can point me in the right direction to fix this issue.. I just recently have a need to have a group of people (5 to be exact) talk via a conference call on a semi-regular basis. The phone lines that are connected to a conference (meetme) are as such: 1. Local SIP (Analog phone using SPA-2000) -- ME 2. Remote SIP (Analog Phone using SPA-2000) via Internet 3. Standard Analog Line via clone X100P card 4. Standard Analog Line via Clone X100P card 5. VOIP Phone line (Broadvoice) via Internet All 5 lines work perfectly fine, and when any two lines connect to each other, both parties hear the other (save occasional Internet blips with Broadvoice) perfectly fine, volume is correct, etc.. When all 5 meet in the conference, I (Phone #1, local SIP) can hear everyone fine, however: The Analog callers (#3 #4) say that they can hear each other, and me just fine, but the Remote Sip (2) and Broadvoice (5) are very low volume and can barely hear them. Conversly, the Remote SIP and Broadvoice callers say they hear Local SIP (Me) and each other just fine, but the Analog callers are very soft and can barely hear them. Like I said, the analog lines are used all the time, and I've tweaked the TX/RX gain so they sound normal during regular phone calls.. Is there some place I can tweak their Gains for Meetme? Or, is there something I can do to the SIP callers to raise their gain while in the conference? And, of course, the really strange thing is that I (Local SIP, #1) hear everybody perfectly.. Thanks for any ideas/suggestions you can give me, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile Error chan_sccp-20050705 on asterisk 1.0.9 (tarball)
Holger Hornung ha scritto: Hello! I tried to compile chan_sccp-20050705 but I receive the following errors: What is the problem? rm /usr/include/asterisk/* rm /usr/lib/asterisk/modules/* cd asterisk make clean make upgrade cd chan_sccp-20050705 make clean make install asterisk -vvvcg Regards, Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some refer transfer questions / issues!
Hello, I think there maybe an issue with my refer transfers. See below or attached: No. TimeSourceDestination Protocol Info 1 0.00192.168.1.2 192.168.1.5 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 2 0.027008192.168.1.5 192.168.1.2 SIP Status: 180 Ringing 3 0.140889192.168.1.5 192.168.1.2 SIP/SDP Status: 200 OK, with session description 4 0.141303192.168.1.2 192.168.1.5 SIP Request: ACK sip:[EMAIL PROTECTED] 5 19.690281 192.168.1.5 192.168.1.2 SIP Request: REFER sip:[EMAIL PROTECTED] 6 19.690602 192.168.1.2 192.168.1.5 SIP Status: 202 Accepted 7 19.690691 192.168.1.2 192.168.1.5 SIP/sipfrag Request: NOTIFY sip:192.168.1.5, with Sipfrag(SIP/2.0 200 OK) 8 19.690722 192.168.1.2 192.168.1.5 SIP Request: BYE sip:192.168.1.5 9 19.691225 192.168.1.5 192.168.1.2 SIP Status: 200 OK 10 20.699733 192.168.1.2 192.168.1.5 SIP/sipfrag Request: NOTIFY sip:192.168.1.5, with Sipfrag(SIP/2.0 200 OK) 11 20.700083 192.168.1.5 192.168.1.2 SIP Status: 481 Call Leg/Transaction Does Not Exist As you can see Asterisk excepts the REFER, Status: 202 Accepted is returned. However believe the NOTIFY is incorrect. If I understand RFC 3515 correctly, the NOTIFIY should respone with 100 Trying, not 200 OK. See http://rfc.arogo.net/rfc3515.html section 4.1 Prototypical REFER callflow. Am I correct, or out in left field? PB No. TimeSourceDestination Protocol Info 1 0.00192.168.1.2 192.168.1.5 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 2 0.027008192.168.1.5 192.168.1.2 SIP Status: 180 Ringing 3 0.140889192.168.1.5 192.168.1.2 SIP/SDP Status: 200 OK, with session description 4 0.141303192.168.1.2 192.168.1.5 SIP Request: ACK sip:[EMAIL PROTECTED] 5 19.690281 192.168.1.5 192.168.1.2 SIP Request: REFER sip:[EMAIL PROTECTED] 6 19.690602 192.168.1.2 192.168.1.5 SIP Status: 202 Accepted 7 19.690691 192.168.1.2 192.168.1.5 SIP/sipfrag Request: NOTIFY sip:192.168.1.5, with Sipfrag(SIP/2.0 200 OK) 8 19.690722 192.168.1.2 192.168.1.5 SIP Request: BYE sip:192.168.1.5 9 19.691225 192.168.1.5 192.168.1.2 SIP Status: 200 OK 10 20.699733 192.168.1.2 192.168.1.5 SIP/sipfrag Request: NOTIFY sip:192.168.1.5, with Sipfrag(SIP/2.0 200 OK) 11 20.700083 192.168.1.5 192.168.1.2 SIP Status: 481 Call Leg/Transaction Does Not Exist___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel configuration for Argentina
Cual es el problema en Argentina? La diferencia deberia ser la señalización unicamente. El resto no cambia. Nosotros usamos TP410 sin problemas pero con DS1 no E1. Saludos, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Monday, July 11, 2005 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Zaptel configuration for Argentina I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is the following: /etc/zaptel.conf loadzone=us defaultzone=us # The first Zaptel card has the FXS port. fxoks=1 fxsks=2-8 /etc/asterisk/zapata.conf [channels] rxgain=0.0 txgain=0.0 musiconhold=default busydetect=yes busycount=5 callprogress=yes echocancel=yes ; Internal FAX machine (FXS #1) signalling=fxo_ks language=en immediate=no callwaiting=yes context=pstn-outbound-fax channel = 1 ; Fax phone line (FXO #8) signalling=fxs_ks language=en group=2 callerid=asreceived context=pstn-inbound-fax channel = 8 ; Voice phone lines (FXO #2, #3, #4, #5, #6, #7) signalling=fxs_ks language=en callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 group=1 useincomingcalleridonzaptransfer=yes callerid=asreceived context=pstn-inbound-voice channel = 2-7 Thanks -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 NOTIFY syntax
I'm rolling out an installation with snom 360s in the near future.
Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a
snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I
have the 360's set up to subscribe and notify for the line use lights,
which works like a charm for interoffice calling (between the 360's,
anyway. The IAXy, 200 and, softphone will be used by less phone
dependant types) but what I can't figure out from the Wiki is if it's
possible to have the ZAP lines notify for the outbound lines so we can
see how many lines are in use.
My configuration looks something like this:
sip.conf:
[mjg]
type=friend
username=mjg
context=sip
callerid=Masuo 6001
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6001
subscribecontext=sip
[pjf]
type=friend
username=pjf
context=sip
callerid=Patrick 6003
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6003
subscribecontext=sip
360 configuration:
fkey6!: dest lt;sip:[EMAIL PROTECTED];user=phonegt;
fkey7!: dest lt;sip:[EMAIL PROTECTED];user=phonegt;
extensions.conf:
[macro-oneline]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
exten = 6001,hint,SIP/mjg
exten = 6001,1,Macro(oneline,${MJG})
exten = 6003,hint,SIP/pjf
exten = 6003,1,Macro(oneline,${PJF})
Is there any convenient way to monitor the status of my FXO lines from
the phones? Or do I have to set up the interested parties with gastman?
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[Asterisk-Users] RTP traffic
Hello. How can I check if the RTP traffic between two channels is bypassed? Some * console command? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk @ Home Voicemail
What happens if you press *98 and enter the extension and password? are you using speakerphone? tried it with the handset only? - Original Message - From: [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Monday, July 11, 2005 7:51 AM Subject: [Asterisk-Users] Asterisk @ Home Voicemail OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B interface to POTS lines. Everything seems to be working just fine, but I have some questions on how to access voicemail options. I can leave a message for an extension, but when I try to retrieve it by using *97 it asks for the password and even though I type in the same password I gave the extension for vmail, it tells me the password is incorrect. I would also like some ideas on how to personalize the greetings for vmail as well. I am a bit of a newbie, so go easy... :-) Thanks, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT- USA reseller list required
Ive got a project where I need to sell a voip QOS product from Australia to US resellers. I dont suppose anyone here knows where I can find a list of a whole heap of US resellers do you in either VOIP or IP space? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help !!! astcc
___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uniden UIP 200 and Asterisk.
Adam, I've tried both the [heath] heading and the [31521] heading. I figure the 31521 was right because the registration error message says [EMAIL PROTECTED] I've tried host = dynamic and defaultip = 172.x No combination of those above settings scores me a successful registration. h From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Edwards Sent: Tuesday, July 05, 2005 5:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uniden UIP 200 and Asterisk. Unless I'm very much mistaken you want to get rid of either the host=dynamic or the defaultip=something host=dynamic indicates the device is getting an IP from dhcp and it will tell * what it is when it registers. defaultip=something indicates that the device is staticip. Devices like this are normally dynamic so try losing the defaultip entry cheers Mark On 7/6/05, Heath Oderman [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk.I've tried a few different settings.I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone is 172.28.184.105 In sip.conf - [uip200] username = heath secret = happy type = friend qualify = no host = dynamic defaultip = 172.28.184.105 dtmfmode = rfc2833 context = sip nat=no In unidenMAC.txt - # Sip Settings MyLcdDisplay 31521 MyDialNumber 703XXX DisplayName31521 UserNameForProxy heath PasswordForProxy happy UserNameForRegistrar heath PasswordForRegistrar happy The output from asterisk is, of course: *CLI Jul4 15:33:15 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from ' sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Jul4 15:33:45 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Jul4 15:34:15 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' I've tried a few variants without much luck.Any pointers would be greatly appreciated. Thanks in advance for any help you can offer. heath Transparent Logic Technologies Heath Oderman 757-410-2593 x 113 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- regards, Mark P. Edwards FWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Peter Nixon to Speak at Cluecon
Peter Nixon will be making the trip to Chicago to speak at Cluecon, he'll be speaking on the topic of Real world deployment of Open Source. Peter has done tremendous amounts of work on the FreeRadius project. In addition if you're wanting to get sponsorship in this is the week to do so, we are sending everything off to the printer to get printed by friday. Thanks, Brian West ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-Dev] Peter Nixon to Speak at Cluecon
Peter Nixon will be making the trip to Chicago to speak at Cluecon, he'll be speaking on the topic of Real world deployment of Open Source. Peter has done tremendous amounts of work on the FreeRadius project. In addition if you're wanting to get sponsorship in this is the week to do so, we are sending everything off to the printer to get printed by friday. Thanks, Brian West ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP traffic
Pepe Aracil wrote: Hello. How can I check if the RTP traffic between two channels is bypassed? Some * console command? You can't. show channels and sip show channels will only show you the SIGNALING (which always passes thru Asterisk). You will need to use tcpdump or etherreal or something like that on one of the Asterisk boxes to find out what you want to know. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video phone settings???
Ronald Wiplinger wrote: apenon apenon wrote: Yes I have faced with the same problem, try to upgrade your eyebeam, some old versions have problem. How to make the echo test? Just add a line to your extensions.conf: exten = 600,1,Echo() And that should do it. Also try the hardphones with different resolutions/bandwidths (CIF/QCIF). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NoOp
It's a little odd. Something like asterisk -v4 seems more appropriate. You can also use set verbose level so that you don't have to restart your console session to change the verbosity. I really don't know what the maximum effective verbose level is. MARK. George Garvey wrote: On Sun, Jul 10, 2005 at 09:49:37PM -0400, MF Hulber wrote: Maybe it shows up after a certain verbosity level. Try asterisk -r When I do that NoOps always show up. Looks like you're right. Guess I never used enough v's ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forward the ALERT_INFO
Is asterisk able to forward it's ALERT_INFO data to another asterisk server ? My situation should look like the following: Call comes into asterisk1 in SIP. Asterisk1 sets the ALERT_INFO=Bellcore-r2, Asterisk1 dials Asterisk2 (SIP), Asterisk2 dials our SIP device which should ring with the Bellcore-r2 Any way to pass the ALERT_INFO through to the SIP device? Thanks -- Benjamin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF not sending properly via IAX
I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits. Any ideas? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF not sending properly via IAX
Before to anything else, are you sending DTMF in-ound or out-bound? Most of the time when DTMF is not sent is because is in-bound. Just choose out-bound or RFC2833 (I don't remember if this is the right standard). Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, July 11, 2005 4:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DTMF not sending properly via IAX I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits. Any ideas? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE and callerID
I've been experimenting with the zaptel TDMoE stuff and I've got it all working. Calls go from one asterisk box to the other, with no issues, except they don't bring the callerID along with them. I tried the em signalling from the wiki and I thought maybe that had something to do with it, so I just changed it to half fxsks and fxoks and that didn't help me any, I still don't get the callerID of the caller, even if I define it in the outgoing end of the TDMoE in zapata.conf So, is there a trick to it or does callerID information just not go across TDMoE? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT- USA reseller list required
I don't suppose anyone here knows where I can find a list of a whole heap of US resellers do you in either VOIP or IP space? This might help: http://www.voip-info.org/tiki-index.php?page=Asterisk+system+vendors -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
You need to upgrade to a older version first. version 5 or 6 before upgrading it to version 7. I appreciate the response but that's what isn't working. I have tried v5.3 and v3.0 with the same result. I suspect the firmware version (P003AM30) is the problem as I haven't run across any Cisco firmware matrix that references this version of the firmware. If anyone has run into this and resolved it I sure would appreciate some clues. -- Marc H. Fishman OuttaSite Resources [EMAIL PROTECTED] If you woke up breathing, congratulations! You have another chance! PGP KeyID: 6C8E212E75CDBD79 PGP Key Fingerprint: E620 1F11 D3AC 6FEC 4CC5 8CA6 6C8E 212E 75CD BD79 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DTMF not sending properly via IAX
Carlos Alperin [EMAIL PROTECTED] wrote: Before to anything else, are you sending DTMF in-ound or out-bound? IAX always sends DTMF out-of-band, not inband. Most of the time when DTMF is not sent is because is in-bound. Just choose out-bound or RFC2833 (I don't remember if this is the right standard). That would be true for SIP, but I am using IAX. The IAX2 debug log shows the DTMF out-of-band packets being sent. Cheers Tony Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, July 11, 2005 4:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DTMF not sending properly via IAX I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits. Any ideas? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: passing through MWI info from SBC
Well, it looks like there is no way for Asterisk to read the MWI fsk tones from the PSTN at this time. Sigh... Anyways, I am going to get a couple outboard boxes with MWI indicators on them, and see if my wife can deal with that to tell if a message is waiting instead of the MWI on the handset. Does anyone here have some recommendations for such devices, preferably with large MWI's? I have plenty of CAT5 jacks in the house, so running an analog signal around next to the ethernet that feeds the VOIP phones is easy. I have seen some caller-ID products that have this, but it's a pretty small MWI on the ones I have looked at. I'd like something like the MWI that's on the existing Nortel ventures - big, bright flashing red if lit. :-) Thanks, Mike --- John Novack [EMAIL PROTECTED] wrote: Mike Myers wrote: John Novack wrote: Mike Myers wrote: Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all exchange voicemails with each other on the SBC voicemail system. They can leave messages for each other without having the phones ring, etc... We have a 2 yr old at home, and her sister has some small kids too, so that's how they manage to send voicemails when they are unsure if the kids are sleeping, etc... Anyway, preserving this capability of using the SBC VM and being notified when a message is waiting is critical for good WAF. The vonage line and voipjet line can be intergrated into the Asterisk VM. My Nortel venture phones light the MWI if any line has VM on it, and the display tells you which lines have VM waiting. I would love to be able to duplicate this function on the Polycom's and hopefully the Sipura's as well. I've looked for answers on this, but haven't found one, hence the post. My apologies if I have missed something. Thanks much, Mike You haven't missed much. With SBC you are out of luck, since Asterisk doesn't detect dialtone ( it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems. Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF not sending properly via IAX
Perhaps we need to go over this again. IAX2 CANNOT DO INBAND DTMF. IAX2 DOES NOT USE RTP. IAX2 DOES NOT DO RFC2833. Carlos Alperin wrote: Before to anything else, are you sending DTMF in-ound or out-bound? Most of the time when DTMF is not sent is because is in-bound. Just choose out-bound or RFC2833 (I don't remember if this is the right standard). Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, July 11, 2005 4:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DTMF not sending properly via IAX I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits. Any ideas? Cheers Tony -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDMoE and callerID
I don't notice it on my TDMoE that is configured as PRI either. Looks like you need to post a bug to the tracker. MATT--- -Original Message- From: Weezey [mailto:[EMAIL PROTECTED] Sent: Monday, July 11, 2005 4:33 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDMoE and callerID I've been experimenting with the zaptel TDMoE stuff and I've got it all working. Calls go from one asterisk box to the other, with no issues, except they don't bring the callerID along with them. I tried the em signalling from the wiki and I thought maybe that had something to do with it, so I just changed it to half fxsks and fxoks and that didn't help me any, I still don't get the callerID of the caller, even if I define it in the outgoing end of the TDMoE in zapata.conf So, is there a trick to it or does callerID information just not go across TDMoE? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE and callerID
Weezey wrote: So, is there a trick to it or does callerID information just not go across TDMoE? Use PRI signaling on the TDMoE span, not quasi-analog signaling. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NoOp
MF Hulber wrote: It's a little odd. Something like asterisk -v4 seems more appropriate. You can also use set verbose level so that you don't have to restart your console session to change the verbosity. I really don't know what the maximum effective verbose level is. MARK. 255 JN George Garvey wrote: On Sun, Jul 10, 2005 at 09:49:37PM -0400, MF Hulber wrote: Maybe it shows up after a certain verbosity level. Try asterisk -r When I do that NoOps always show up. Looks like you're right. Guess I never used enough v's ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
Answering my own question here for anyone else fighting with this. From a Cisco Field Notice (see http://www.cisco.com/warp/public/770/fn18246.shtml). The problem appears to be that a 7960/7940 running P003AM30, the load shipped from the factory, cannot load a new load file that is more than 393216 bytes in size. This corresponds to 768 TFTP packets of 512 bytes each. Once the phone receives the 769th packet, it will send an error to the TFTP server indicating Disc full or allocation exceeded - error code 3. It appears that this phone load cannot handle a file larger than 384Kb. -- Marc H. Fishman OuttaSite Resources [EMAIL PROTECTED] If you woke up breathing, congratulations! You have another chance! PGP KeyID: 6C8E212E75CDBD79 PGP Key Fingerprint: E620 1F11 D3AC 6FEC 4CC5 8CA6 6C8E 212E 75CD BD79 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc / incoming call - error 6
Hi Folks,
I've Asterisk Bristuffed up and running behind an Auerswald Commander
Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works
marvelleous for outgoing calls (as the parallely installed avm fritzcard
with chan_capi does), but when I'm trying to call in, I get a short ring
signal and then the connection is terminated. This does not happen with
chan_capi and the avm card though. I've configured the extension 500
on the internal bus under which the card should be available.
I couldn't figure out what causes the trouble tough. Thanks for your help!
yours,
Alexander
The relevant section from the bri debug span 1 on the asterisk console
is the followin:
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
Request
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 1 (reference 197/0xC5) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 81 90]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
Jul 11 22:50:05 WARNING[13679]: chan_zap.c:7504 zt_pri_error: PRI: !!
Got reject for frame 29, but we have nothing -- resetting!
#
Here's my configuration and the full trace of the bri activity.
;ZAPHFC Konfiguration
switchtype = euroisdn
;signalling = bri_net_ptmp ;this is for a peer to multipeer network
signalling = bri_cpe_ptmp
pridialplan=local
prilocaldialplan=local
;pritrustusercid = yes
overlapdial=yes
language=de
immediate=no
group = 1
context=gtiin
;echocancel=yes
channel = 1-2
;extension.conf configuration
[gtiin]
exten = 500,1,SetCallerId(${CALLERIDNUM})
exten = 500,2,SetCIDName(${CALLERIDNAME})
exten = 500,3,Dial(IAX2/alexSIP/alex,60,r)
exten = 500,4,Hangup
[gtiout]
;section for outgoing calls via prefix 0
exten = _0.,1,SetCallerID(${CALLERIDNUM})
exten = _0.,2,Dial(ZAP/g1/${EXTEN},60,Ttr)
exten = _0.,3,Hangup
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g currently running on
75 (pid =
7032)
Verbosity is at least 35
75*CLI bri debug span 1
Enabled debugging on span 1
Protocol Discriminator: Q.931 (8) len=39
Call Ref: len= 1 (reference 69/0x45) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
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: A-Law (35)
[18 01 89]
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive Dchan: 0
ChanSel: B1 channel
]
[6c 0f 21 80 30 30 34 33 32 32 34 33 33 35 33 31 36]
Calling Number (len=17) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '0043xx' ]
[70 04 81 35 30 30]
Called Number (len= 6) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '500' ]
[7d 02 91 81]
IE: High-layer Compatibility (len = 4)
-- Making new call for cr 69
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 125 (cs0, High-layer Compatibility)
Protocol Discriminator: Q.931 (8) len=11
Call Ref: len= 1 (reference 197/0xC5) (Terminator)
Message type: SETUP ACKNOWLEDGE (13)
[18 01 89]
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive Dchan: 0
ChanSel: B1 channel
]
[1e 02 81 82]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Private network serving the local user (1)
Ext: 1 Progress Description: Called
equipment is non-ISDN. (2) ]
-- Executing SetCallerID(Zap/1-1, 0043x) in new stack
-- Executing SetCIDName(Zap/1-1, 0043xx) in new stack
-- Executing Dial(Zap/1-1, IAX2/alexSIP/alex|60|r) in new stack
-- Called alex
Jul 11 22:49:55 NOTICE[13679]: app_dial.c:759 dial_exec: Unable to
create channel of type 'SIP'
-- Accepting voice call from '0043xxx' to '500' on channel 0/1,
span 1
Protocol Discriminator: Q.931 (8) len=7
Call Ref: len= 1 (reference 197/0xC5) (Terminator)
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