Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
AGX-Addons crashes Asterisk for us. Working solution (on 100+ servers we installed): - apt-get -y install g++ libtiff4 libtiff4-dev patch autoconf automake libtiff-tools cd /usr/src wget http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20080402.tar. gz tar xzvf spandsp-20080402.tar.gz cd /usr/src/spandsp-0.0.4 ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig cd /usr/src wget http://193.138.191.205/packets/fax_apps_asterisk14.tgz tar xzvf fax_apps_asterisk14.tgz cd /usr/src/fax_apps make make install Restart Asterisk. Voila! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Sent: Wednesday, April 16, 2008 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing No progress at all. Version from Debian/Lenny repository still crashes and I'm not able to compile AGX. It gives out a long list of error messages. Some unsatisfied dependencies...? I Can't experiment for a while after unwanted night-time visit of fire-fighters :-( I have to let everything dry and clean out of sand and drywall pieces :-( Martin - Original Message - From: Justin Newman mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: 11. dubna 2008 13:00 Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing Did this just start happening with the 1.4 tree? Have you made any progress on getting it resolved? Justin Newman Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: Let's be more specific here, folks: What version numbers? Asterisk, spandsp, agx-addons / rx-tx-fax? Asterisk: yesterday's 1.4 SVN SpanDSP: tried with pre 15, 16 and 18 AGX-Addons: tried with 1.4.5 and svn trunk rx/txfax: supplied by AGX Addons - although they seem to build the files and stick them into the modules directory, rather than adding to the apps directory and modifying the Makefile. i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 search=0xD6506D20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Lite and Presence?
Cool - thanks Rob. I will check it out tmorrow. Simon On Wed, Apr 16, 2008 at 4:34 PM, Rob Hillis [EMAIL PROTECTED] wrote: IIRC Asterisk doesn't support the full presence publishing spec so you won't get the full range of possible status types, however you should at least get free/busy. I vaguely recall having to change the presence type from peer-to-peer to something else - that's done in the SIP configuration window. However, since I don't have X-Lite in front of me at the moment (fortunately, for the most part!) I can't give you more of a hint than that. Simon wrote: Thanks again!.. Right. I have it working now, it shows the users statuses as online or offline and changes them when someone closes their app. But not free/busy type changes.. Any idea why here? Simon On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote: X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:48057d5e261007514015341! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with B410P
Hello, Let me ask for help on a problem That I can not solve a 2 B410P on my server I can not mounted ports TE UP Everything seems to have been successfully compile no apparent errors Misdn (1_1_7_2 version) Zaptel (version 1.4.9.2) Asterisk (version 1.4.18) Kernel (version 2.6.17-5mdv) Misdn-init scan Card=1.0x4 Card=2.0x4 MISDN scan BN4S0 BN4S0 Extract from misdn.log Tue Apr 15 14:39:34 2008: P [0] - mISDN Channel Driver Registered -- Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [1] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [2] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [3] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [4] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [5] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [6] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [7] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:34 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:34 2008: P [8] MGMT: SSTATUS: L2_RELEASED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED Tue Apr 15 14:39:41 2008: P [0] MGMT: SSTATUS: L1_DEACTIVATED # Test call # Tue Apr 15 14:40:22 2008: P [0] Checking Port: 0 Tue Apr 15 14:40:22 2008: P [0] - Group Call group: first_extern Tue Apr 15 14:40:22 2008: P Group [1] [first_extern] Port [1] Tue Apr 15 14:40:22 2008: P [1] PMP down Port Tue Apr 15 14:40:22 2008: P [1] portup: 0 Tue Apr 15 14:40:22 2008: P Group [2] [first_extern] Port [2] Tue Apr 15 14:40:22 2008: P [2] PMP down Port Tue Apr 15 14:40:22 2008: P [2] portup: 0 Tue Apr 15 14:40:22 2008: P Group [3] [first_extern] Port [3] Tue Apr 15 14:40:22 2008: P [3] Port down PMP Tue Apr 15 14:40:22 2008: P [3] portup: 0 Tue Apr 15 14:40:22 2008: P Group [4] [first_extern] Port [4] Tue Apr 15 14:40:22 2008: P [4] PMP down Port Tue Apr 15 14:40:22 2008: P [4] portup: 0 Tue Apr 15 14:40:22 2008: P [5] Group [first_extern] Port [5] Tue Apr 15 14:40:22 2008: P [5] PMP down Port Tue Apr 15 14:40:22 2008: P [5] portup: 0 Tue Apr 15 14:40:22 2008: P [6] Group [first_extern] Port [6] Tue Apr 15 14:40:22 2008: P [6] Port down PMP Tue Apr 15 14:40:22 2008: P [6] portup: 0 Tue Apr 15 14:40:22 2008: P [7] Group [first_extern] Port [7] Tue Apr 15 14:40:22 2008: P [7] PMP down Port Tue Apr 15 14:40:22 2008: P [7] portup: 0 Tue Apr 15 14:40:22 2008: P [8] Group [first_extern] Port [8] Tue Apr 15 14:40:22 2008: P [8] PMP down Port Tue Apr 15 14:40:22 2008: P [8] portup: 0 Asterisk log extract P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED P [0] MGMT: SSTATUS: L1_DEACTIVATED * CLI misdn show stacks BEGIN: STACK_LIST * Type 1 TE Prot. PMP L2Link DOWN L1Link: DOWN Bloked: 0 Debug: 4 * Type 2 TE Prot. PMP L2Link DOWN L1Link: DOWN Bloked: 0 Debug: 4 * Type 3 TE Prot. PMP L2Link DOWN L1Link: DOWN Bloked: 0 Debug: 4 * Type 4 TE Prot. PMP L2Link DOWN L1Link: DOWN Bloked: 0 Debug: 4 So Do you have an idea With my thanks Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] do cards just instantly go bad
In article [EMAIL PROTECTED], Jerry Geis [EMAIL PROTECTED] wrote: Hi - Been using a TE205P for a number of months - no issues. Today I was talking to someone and I heard click No more phone service. I still have data service on this T1 line. (partial phone) zttool reports the SPAN as OK. calls are not coming in or going out. Does a card just go bad like that? How can I tell if the card is bad? I was expecting/hoping to see something other than OK on zttool. Its reporting OK but still no calls. I made no changes to anything in weeks. I presume there is a chance the carrier (nuvox) is having issues but how can I make sure there isnt something on my end? Since you have two T1 ports, make up a T1 crossover cable (pair 1-2 crossed with pair 4-5) and connect the ports together. In /etc/asterisk/zapata.conf set one of the spans to signalling=pri_cpe and the other to signalling=pri_net. Send one of the spans to a test context that just defines extension _X! and plays the demo message or something. Then try dialling out on the other span. If you've done everything right and it doesn't work, it's the card. If it does work, then your problem is with the carrier. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple queue announcements
Greetings list, I've been playing around with queues on an old asterisk 1.2 box at a customer's site. They want to be able to add really simple queue announcements every minute, along the following lines: sorry for the delay, someone will be with you shortly. Looking at the announce options in queues.conf, it seems to be possible to announce queue position and/or hold time, but not a simple announcement. I suppose I could, theoretically, erase all the queue-youarenext, queue-callswaiting, etc. voice files, but that seems rather excessive to achieve something so simple. Any thoughts/suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple queue announcements
use the option periodic-announce On Wed, Apr 16, 2008 at 1:47 PM, Chris Bagnall [EMAIL PROTECTED] wrote: Greetings list, I've been playing around with queues on an old asterisk 1.2 box at a customer's site. They want to be able to add really simple queue announcements every minute, along the following lines: sorry for the delay, someone will be with you shortly. Looking at the announce options in queues.conf, it seems to be possible to announce queue position and/or hold time, but not a simple announcement. I suppose I could, theoretically, erase all the queue-youarenext, queue-callswaiting, etc. voice files, but that seems rather excessive to achieve something so simple. Any thoughts/suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple queue announcements
Chris Bagnall wrote: Greetings list, I've been playing around with queues on an old asterisk 1.2 box at a customer's site. They want to be able to add really simple queue announcements every minute, along the following lines: sorry for the delay, someone will be with you shortly. I believe you'll need to migrate them to 1.4. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 480 Do Not Disturb
Hi, Johansson Olle E wrote: Well, 480 translates to AST_CAUSE_NOANSWER - cause 19 - check by checking HANGUPCAUSE instead of DIALSTATUS and you will get many more details. Great, that's all I need: It gives me more ways to analyse the different reason for the hangup and I can use the different numbers to return different explanations with Playback(). My client doesn't only want to hear the busy signal, but wants to play a file with an explanation why the call couldn't be established. Thanks again, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with Asterisk 1.4.19 -accountcode dissapearing
Thanks, that`s what I ended up doing. Still, it doesn't seem to be WAD,
since the CDR(accountcode) is correct and suddently dissapears.
Is this a bug (I was looking through the bug system and couldnt match this
with a bug, but then again I am not a developer) or is it really WAD?
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mindaugas
Kezys
Sent: Tuesday, April 15, 2008 08:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] problem with Asterisk 1.4.19 -accountcode
dissapearing
As far as I noticed this issue is not 1.4.19 only. Same thing happens on
all Asterisk versions.
Set your own variable before transfer:
Exten = , Set(__MYACC=${CDR(accountcode)})
And use ${MYACC} in other (transfered) calls.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, April 15, 2008 3:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] problem with Asterisk 1.4.19 - accountcode
dissapearing
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS CORRECT AND IS
EQUALS TO THE ACCOUNTCODE SPECIFIED MUCH EARLIER IN THE DIALPLAN
exten = _X!.,n,Gotoif($[${i} = 1]?$[${PRIORITY}+2])
;DIAL ALL MAC PHONE ASSOCIATED WITH THIS EXTENSION SIMULATENOUSLY
exten =
_X!.,n,Dial(${mac_dial_string:0:$[${LEN(${mac_dial_string})}-20]}|${sip_phon
es_ring_time}) ;remove least 7 characters, thos
e are left there by the invalid last SQL fetch
exten = _X!.,n,Set(i=0)
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS EMPTY, and so
is this variable if I use it in any way.
When I dial an extension and it hits this diaplan, it works fine. But if I
dial an extension, answer and then transfer (using Polycom phones) to an
extension using this dialplan I lose the accountcode where specified in the
code. It's empty. How can ${CDR(accountcode)} lose it's value for no
reason in those two seemingly innocent diaplan lines?
Below is the CLI output if it's useful:
-- Executing [EMAIL PROTECTED]:22]
NoOp(SIP/0004f2134384-1-097fb4e8, 1234567890) in new stack ;THIS IS THE
ACCOUNTCODE
-- Executing [EMAIL PROTECTED]:23]
GotoIf(SIP/0004f2134384-1-097fb4e8, 0?25) in new stack
-- Executing [EMAIL PROTECTED]:24]
Dial(SIP/0004f2134384-1-097fb4e8, SIP/0004f2134384-3|8) in new stack
-- Called 0004f2134384-3
-- SIP/0004f2134384-3-099947b0 is ringing
== Spawn extension (generic-extensions-db, 705, 24) exited non-zero on
'SIP/0004f2134384-1-097fb4e8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back from
192.168.1.6
-- Nobody picked up in 8000 ms
-- Executing [EMAIL PROTECTED]:25]
Set(SIP/0004f212ae63-1-099700a8, i=0) in new stack
-- Executing [EMAIL PROTECTED]:26]
NoOp(SIP/0004f212ae63-1-099700a8, ) in new stack ;MISSING ACCOUNTCODE
IS HERE
Mick
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[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Hello Asterisk People, I have two annoying bugs in asterisk, that i want to know if some of you have already found a way to fix: Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch. 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Not so nice thing to do 2. AgentCallBack I know i shouldn't have to use this function, since it is deprecated but lets comment the behavior Everything works fine, but when there are calls waiting in the queue, and the agent log in using this function, the agent is able to take the call , but the system log off immediately after the agent hang up the call. No solution at the moment, just login in and log in until there are no waiting calls, for the agent to not be kicked off. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup conundrum with RxFAX
Heres something that's making me scratch my head... I'm using RxFAX on ISDN lines and in-general it's going well. However, there seems to be a case when the fax doesn't get delivered, but looking through the CDRs it seems that the call happened, RxFAX was executed .. time passed (1-2+ minutes) then hangup. I'm wondering if some FAX machines just hangup after the call rather than complete some sort of ending negotiation, or if the RxFAX part misses the end and just sees the hangup.. Now, in a normal fax machine, it's going to print the fax regardless, even if the last page is only half full because of a genuine line drop or hangup, but it seems that: [Description] RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the ... Returns -1 when the user hangs up. Returns 0 otherwise. So if it's returning -1, then the call/channel is hungup, and any dialplan instructions after it won't get executed, even though there might be some (or all) pages of the fax sitting in the receive file... Does this make sense to anyone, or am I barking up the wrong tree! My thoughts now are to actually do a hangup at the end of the RxFAX and rely on a 'h' extension to pick it up and carry on with the 2nd half (which is PDFing and emailling the fax), but I'm concerned I'm going to lose the channel variables as it suggests on the wiki, so I'll lose the REMOTESTATIONID string and caller ID... Anyone with any experience of this, or suggestions otherwise? Thanks, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Interface Status Bug and Re: Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
I forget another bug, i use the asterisk manager interface. I frequently use the status function but it doesn't work as expected, i use a program to parse the output of the status command but it don't behave as expected, because i always wait for the latest package: StatusComplete, and this package never arrives in the stream sometimes StatusComplete is shown, sometimes not, but when it decide not to show StatusComplete, asterisk really don't show StatusComplete, so sad for me... Temporarily workaround: put a timeout on the socket read function to assume the asterisk manager is not working properly. Well if anybody have found some solution to this i will appreciate your comments. Slds. Nestor A. Diaz wrote: Hello Asterisk People, I have two annoying bugs in asterisk, that i want to know if some of you have already found a way to fix: Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch. 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Not so nice thing to do 2. AgentCallBack I know i shouldn't have to use this function, since it is deprecated but lets comment the behavior Everything works fine, but when there are calls waiting in the queue, and the agent log in using this function, the agent is able to take the call , but the system log off immediately after the agent hang up the call. No solution at the moment, just login in and log in until there are no waiting calls, for the agent to not be kicked off. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Version FIOS MWI Detection - asterisk-1.6-beta7
I'm trying to get the Telco MWI recognition working in asterisk-1.6-beta7. I'm told that it's supposed to work provided my telco support FSK MWI signalling. I have Verzon FIOS. I believe I have FSK MWI signaling as I can hear the standard stutter tone when I pick up a live handset in front of my asterisk connection. I have the Verizon line attached to a TDM400 card, port 4. I have these in my zapata.conf file: ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes mwilevel=256 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming channel = 4 Does anyone have this feature working? Do you see anything wrong with my configuration? Thanks, Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
We have two servers but looks like G729 issues. Works fine on the old server and not sure if it is T1 related. See SIP Debug. Any experiences to share. Thanks --- Newark1*CLI --- SIP read from 194.xx.Xx.Xx:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=K784d2637;rport From: Cell Phone DC sip:[EMAIL PROTECTED];tag=as04819ca3 To: sip:xx;tag=xx Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 198 v=0 o=xx 12x 12 IN IP4 62.xx.xx.xx s=SIP Call c=IN IP4 62.xx.xx.xxx t=0 0 m=audio 8786 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 - --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 62.xx.xx.xx:8786 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 62.xx.xx.xx:8786 -- SIP/Voicetrading-08e1ce18 is making progress passing it to Zap/5-1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager Interface Status Bug and Re: Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
On Wed, 2008-04-16 at 07:55 -0500, Nestor A. Diaz wrote: I frequently use the status function but it doesn't work as expected, i use a program to parse the output of the status command but it don't behave as expected, because i always wait for the latest package: StatusComplete, and this package never arrives in the stream sometimes StatusComplete is shown, sometimes not, but when it decide not to show StatusComplete, asterisk really don't show StatusComplete, so sad for me... This sounds like it could be a bug... please open a ticket on the bug tracker (http://bugs.digium.com/) so that the developers can keep track of it and make sure it gets fixed. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Tue, Apr 15, 2008 at 7:07 PM, Shaun Ruffell [EMAIL PROTECTED] wrote: Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Just tried it... Behaviour looks equivalent. Drivers load ok, ztcfg leads to BUG: soft lockup detected on CPU#1... dmesg snippet is: Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-mattf-zaptel-1.4-stackcleanup-r4163M Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 138 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f89c4000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x37407400 Reg 1: 0x37407000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x000200ff Reg 9: 0x00f5 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#1! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f8f6b1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f8f52b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8f56ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8966fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f89945ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f899bf39] zt_ioctl+0x273/0x14be [zaptel] [c045] chrdev_open+0x11e/0x132 [c0477657] chrdev_open+0x0/0x132 [c046e9e6] __dentry_open+0xea/0x1ab [c0604451] schedule+0x90d/0x9ba [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c0470daa] __fput+0x13f/0x167 [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === wcte12xp0: Missed interrupt. Increasing latency to 4 ms in order to compensate. VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 SPAN 3: Secondary Sync Source timing source auto card 0! Completed startup! wcte12xp: Clearing yellow alarm Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set ...as mentioned previously, current kernel has CONFIG_4KSTACKS set. I'll now go ahead and rebuild a kernel with 4K stacks disabled. I'll post back later. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy (congestion) signal and cell phones
Hello, all! I've noticed a peculiar situation and I am hoping someone can shed some light on it for me. We have an Asterisk (1.4.18 ) box talking to the world via Zaptel on a PRI from a telco (USA). I have an extension that returns busy signal (fast-busy or regular busy) (using US tones). When I call from a landline or from another PBX, I get a busy signal, just like I expect. But when I call from a cell phone, the cell phone terminates the call as soon as connection is established. I've tested several cell phone models from different providers in the US. Same thing happens with calls coming from Gizmo. I manually changed the tones I send back (with Playtones) to mimic Austrian busy tone (picked the first one in the list from indications.conf) . Now, from the cell phone and Gizmo alike, I get busy tones. So, my questions is: why do cell phones and Gizmo both detect busy tones and terminate the call? Is that a standard behavior? Why don't landlines do that? Thank you in advance. Regards, Mark G. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi and SIP
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]mailto:SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and pass
the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer a.b.c.d/${NUMBER}
cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip peer
per box(I have 25 connected in my dundi cloud), nor would I like to enable
anonymous SIP calls, as I have the ports open to the world for inbound sip from
bandwidth.com
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Re: [asterisk-users] Problem with B410P
Could be this... http://www.misdn.org/index.php/FAQ_chan_mISDN#Why_does_the_L1_goes_DOWN_on_my_PMP_Isdn_Link.3F_Or_why_do_i_get_No_free_chan_even_after_group_call_from_chan_misdn_if_dialing_out_on_my_PMP_Link.3F Hmmm... that's a long link. It is the Why does the L1 goes DOWN on my PMP Isdn Link FAQ in the chan_misdn FAQ at http://www.misdn.org/index.php/FAQ_chan_mISDN In short, the telcos shut PMP links down to save on power costs, you need to bring them up before initiating an outbound call. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callerid Error
Asterisk Users, I am running a Debian Etch system with Asterisk 1.4.11 with a TDM03B card. Once in awhile, I get this error on the Asterisk, which causes my channels to be busy/congested, leaving me with just one channel to recieve and make calls: NOTICE[31454]: chan_zap.c:6367 ss_thread: Got event 17 (Polarity Reversal)... WARNING[31454]: chan_zap.c:6499 ss_thread: CallerID returned with error on channel 'Zap/3-1' What could be causing this issue? Any would input would be greatly appreciated. Thanks In Advance, John _ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_WL_Refresh_messenger_video_042008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under 2.6.18-53.el5) HW: Digium TE220B, the one with HW echo cancellation (configured as 2x E1 via jumpers) Context: Pre-site installation of system, no E1 conectivity (loopbacks tested) /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=25-39,41-55 dchan=40 span=2,2,0,ccs,hdb3,crc4 bchan=56-70,72-86 dchan=71 Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel buffer: About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#0! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042621c] release_console_sem+0x17e/0x1b8 [c0407406] do_IRQ+0xa5/0xae [f8994311] t4_dacs+0x211/0x24b [wct4xxp] [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel] [c0457600] mempool_alloc+0x28/0xc9 [c04ddd33] cfq_resort_rr_list+0x23/0x8b [c04deb6c] cfq_add_crq_rb+0xba/0xc3 [c04dec72] cfq_insert_request+0x42/0x498 [c04d5175] elv_insert+0x10a/0x1ad [c04d908b] __make_request+0x31d/0x366 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04dde27] __cfq_slice_expired+0x8c/0xa5 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04d505d] elv_next_request+0x15c/0x16a [f88bc101] start_io+0x77/0xdc [cciss] [f88bf63e] do_cciss_request+0x32c/0x337 [cciss] [f88ccff0] __split_bio+0x408/0x418 [dm_mod] [f88cd6a6] dm_request+0xce/0xd4 [dm_mod] [c04d6a81] generic_make_request+0x248/0x258 [c04d8734] submit_bio+0xbf/0xc5 [c04548e2] find_get_page+0x18/0x38 [c04719ad] __find_get_block_slow+0xfb/0x105 [c0471cea] __find_get_block+0x15c/0x166 [c0471cea] __find_get_block+0x15c/0x166 [c0471d24] __getblk+0x30/0x270 [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd] [f885a472] journal_cancel_revoke+0x77/0x96 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c041f871] __wake_up+0x2a/0x3d [f8856679] journal_stop+0x1b0/0x1ba [jbd] [c042a209] current_fs_time+0x4a/0x55 [c048626d] touch_atime+0x60/0x8f [c04552ee] do_generic_mapping_read+0x421/0x468 [c045478b] file_read_actor+0x0/0xd1 [c04548e2] find_get_page+0x18/0x38 [c0457319] filemap_nopage+0x192/0x315 [c046048f] __handle_mm_fault+0x85e/0x87b [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Secondary Sync Source Completed startup! Soft lockup ?! Hmmm... I'm ignorant on this, but it smells fishy ! For completeness sake, driver was previously loaded ok: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 98 Found TE2XXP at base address fdff, remapped to f8854000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x375a2400 Reg 1: 0x375a2000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff2031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) After trying lot's of things (disable ILO, disable USBs, try different kernel, different TE220B, etc), I figured that this soft hangup does not show under zaptel 1.4.9.2... In all due honesty, I haven't got the faintest idea what kind of impact this could have. Side testing zaptel 1.4.10 on a simpler system, an HP Proliant ML110 (nearly a PC), the error does not show up as well. I checked the zaptel 1.4.10 ChangeLog and there are some changes which I'd suspect: 2008-04-01 16:39 + [r4122] sruffell [EMAIL PROTECTED]: * kernel/wct4xxp/base.c: Work
Re: [asterisk-users] CDR and transfers! :(
Hi Raul, CDR's for transfers are beyond the ability of Asterisk. http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html http://bugs.digium.com/view.php?id=11093 It's not something the powers that be want to think about a design for and the solution that's been suggested is to date it to use a different type of server software, such as a SIP Proxy, to generate the CDR's (something easy to suggest and complicated to do). You're not the only one affected by this and there is no fix. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple queue announcements
Doug Lytle wrote: Chris Bagnall wrote: Greetings list, I've been playing around with queues on an old asterisk 1.2 box at a customer's site. They want to be able to add really simple queue announcements every minute, along the following lines: sorry for the delay, someone will be with you shortly. I believe you'll need to migrate them to 1.4. Doug Works fine for me on 1.2.24 ... from queues.conf:- periodic-announce = Custom/periodic-fxqueue01 periodic-announce-frequency=90 plays a brief sorry,etc, press any key to leave a message or continue to hold for next available etc, etc regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy (congestion) signal and cell phones
On Wed, 16 Apr 2008 08:40:42 -0500, Mark Gimelfarb [EMAIL PROTECTED] wrote: why do cell phones and Gizmo both detect busy tones and terminate the call? Is that a standard behavior? It *is* standard procedure for a cellphone to terminate a call immediately it discovers that the called number is busy. It will then, optionally, initiate its auto-redial function etc. Why don't landlines do that? Because back in the old days there were no intelligent electronics to tell the user that the call failed. A special busy tone had to be generated to inform the user that they should hang the receiver up manually. Some traditions die hard. -- Godwin Stewart - Horwich IT services ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy (congestion) signal and cell phones
What country are you in?? Yes, it is common for cell phones to disconnect the call if they receive CONGESTION, but not BUSY. Horwich IT Services (Godwin Stewart) wrote: It *is* standard procedure for a cellphone to terminate a call immediately it discovers that the called number is busy. It will then, optionally, initiate its auto-redial function etc. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best Click-to-call client
Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wcfxo and X100P card won't play nice.
Alex Balashov wrote: Greetings, This may have already been asked many times, but I cannot seem to find a satisfactory and consistent answer anywhere. I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850 or 2650 (cannot recall): 00:00.0 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) (rev 13) 00:00.1 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) 00:00.2 Host bridge: Broadcom CMIC-LE 00:04.0 Class ff00: Dell Embedded Remote Access or ERA/O 00:04.1 Class ff00: Dell Remote Access Card III 00:04.2 Class ff00: Dell Embedded Remote Access: BMC/SMIC device 00:0e.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) 00:0f.0 Host bridge: Broadcom CSB5 South Bridge (rev 93) 00:0f.1 IDE interface: Broadcom CSB5 IDE Controller (rev 93) 00:0f.2 USB Controller: Broadcom OSB4/CSB5 OHCI USB Controller (rev 05) 00:0f.3 ISA bridge: Broadcom CSB5 LPC bridge 00:10.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:10.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 01:06.0 Communication controller: Motorola Wildcard X100P 03:06.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 03:08.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 04:08.0 PCI bridge: Intel Corporation 80303 I/O Processor PCI-to-PCI Bridge (rev 01) 04:08.1 RAID bus controller: Dell PowerEdge Expandable RAID Controller 3/Di (rev 01) But, wcfxo won't recognise it: NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 Failed to initailize DAA, giving up... wcfxo: probe of :01:06.0 failed with error -5 This is running on a custom-compiled kernel 2.6.24.3 with Asterisk 1.4.18 and Zaptel 1.4.10. Any ideas? -- Alex Is that a genuine X100P or a clone? Looks like a modem that has been made to look like an X100P. -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy (congestion) signal and cell phones
I'm in the US, so I was originally using the US tones. Looks like I'm getting a disconnect with both CONGESTION and BUSY. In fact, I wasn't actually using Congestion() and Busy(), I just did Playtones() for both of those. There is no reason to send PRI messages to cell phones, is there? The way I understand, they do frequency interpretation on the incoming tones, just like analog lines do voltage variations. So, to test that, I Playtones()'ed (Pardon my DialPlan-ish dialect) Austrian busy tones--and the cell phone actually played tones back to me. So, to me that means that cell phones look for frequency sequences that they recognize. Now, here's an interesting observation. If I take an analog phone and take it off hook and then call that number from a cell phone, I do hear a busy tone. Is that because analog equipment doesn't generate the exact tone sequence due to analog limitations? This is in addition to my original question. Regards, Mark. What country are you in?? Yes, it is common for cell phones to disconnect the call if they receive CONGESTION, but not BUSY. Horwich IT Services (Godwin Stewart) wrote: It *is* standard procedure for a cellphone to terminate a call immediately it discovers that the called number is busy. It will then, optionally, initiate its auto-redial function etc. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple queue announcements
Drew Gibson wrote: Works fine for me on 1.2.24 ... Sorry, I thought periodic announcements were a 1.4 thing. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wcfxo and X100P card won't play nice.
On Wed, Apr 16, 2008 at 09:44:25AM -0500, Brent Davidson wrote: Alex Balashov wrote: Greetings, This may have already been asked many times, but I cannot seem to find a satisfactory and consistent answer anywhere. I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850 or 2650 (cannot recall): 00:00.0 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) (rev 13) 00:00.1 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) 00:00.2 Host bridge: Broadcom CMIC-LE 00:04.0 Class ff00: Dell Embedded Remote Access or ERA/O 00:04.1 Class ff00: Dell Remote Access Card III 00:04.2 Class ff00: Dell Embedded Remote Access: BMC/SMIC device 00:0e.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) 00:0f.0 Host bridge: Broadcom CSB5 South Bridge (rev 93) 00:0f.1 IDE interface: Broadcom CSB5 IDE Controller (rev 93) 00:0f.2 USB Controller: Broadcom OSB4/CSB5 OHCI USB Controller (rev 05) 00:0f.3 ISA bridge: Broadcom CSB5 LPC bridge 00:10.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:10.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 01:06.0 Communication controller: Motorola Wildcard X100P 03:06.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 03:08.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 04:08.0 PCI bridge: Intel Corporation 80303 I/O Processor PCI-to-PCI Bridge (rev 01) 04:08.1 RAID bus controller: Dell PowerEdge Expandable RAID Controller 3/Di (rev 01) But, wcfxo won't recognise it: NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 Failed to initailize DAA, giving up... wcfxo: probe of :01:06.0 failed with error -5 This is running on a custom-compiled kernel 2.6.24.3 with Asterisk 1.4.18 and Zaptel 1.4.10. Any ideas? -- Alex Is that a genuine X100P or a clone? Looks like a modem that has been made to look like an X100P. What's the difference? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. Matt, Thanks for your feedback. We've already tested the following branch as per Shaun's suggestion, without getting a different behaviour (see today's earlier email to the list): http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? We are now in the middle of rebuilding a non 4K stack page kernel so as to give it a try with 1.4.10, the branch Shaun suggested, 1.4.9.2 and the branch you mention, if it is in fact different from Shaun's. We wait your confirmation and will post non 4K stack kernel results later today. One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Hi Al, Al Baker wrote: Shaun - Could you clarify your post a bit ? 1 - Is the 4 K stacks a Known Problem ? a) If so is it known to be problem on any specific Linux distro ? b) Should ALL installation Check for this PRIOR to doing an Asterisk Install ? I wouldn't really say a known *problem*, since it really depends on what other code is running in the system at the time. I just mentioned that because I've seen 8K stacks help in certain situations. 8K stacks are still the default configuration option in the vanilla kernel. Some distributions (CentOS / Fedora) have switched to 4K by default because they help with memory consumption in highly threaded environments like web servers. For the most part, kernel panics and oops are best handled on a case by case basis with Digium's tech support department since each case is unique. 2) The branch you mention below - are fixes from it in Any current * release ? Not that I'm aware of... Cheers, Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Shaun Ruffell wrote: Hi Al, Al Baker wrote: Shaun - Could you clarify your post a bit ? 1 - Is the 4 K stacks a Known Problem ? a) If so is it known to be problem on any specific Linux distro ? b) Should ALL installation Check for this PRIOR to doing an Asterisk Install ? I wouldn't really say a known *problem*, since it really depends on what other code is running in the system at the time. I just mentioned that because I've seen 8K stacks help in certain situations. 8K stacks are still the default configuration option in the vanilla kernel. Some distributions (CentOS / Fedora) have switched to 4K by default because they help with memory consumption in highly threaded environments like web servers. For the most part, kernel panics and oops are best handled on a case by case basis with Digium's tech support department since each case is unique. In this case, it looks like his kernel is compiled with the softlockup detector code and it is falsely triggering. Disabling that should remove the warning message at the very least. 2) The branch you mention below - are fixes from it in Any current * release ? They will be in the next Zaptel release. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with B410P
Re, That is correct, in the case of France Telecom Pb resolved. With my thanks A + Yves Le mercredi 16 avril 2008 à 14:45 +0100, Ex Vito a écrit : Could be this... http://www.misdn.org/index.php/FAQ_chan_mISDN#Why_does_the_L1_goes_DOWN_on_my_PMP_Isdn_Link.3F_Or_why_do_i_get_No_free_chan_even_after_group_call_from_chan_misdn_if_dialing_out_on_my_PMP_Link.3F Hmmm... that's a long link. It is the Why does the L1 goes DOWN on my PMP Isdn Link FAQ in the chan_misdn FAQ at http://www.misdn.org/index.php/FAQ_chan_mISDN In short, the telcos shut PMP links down to save on power costs, you need to bring them up before initiating an outbound call. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wcfxo and X100P card won't play nice.
Tzafrir Cohen wrote: On Wed, Apr 16, 2008 at 09:44:25AM -0500, Brent Davidson wrote: Alex Balashov wrote: Greetings, This may have already been asked many times, but I cannot seem to find a satisfactory and consistent answer anywhere. I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850 or 2650 (cannot recall): 00:00.0 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) (rev 13) 00:00.1 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) 00:00.2 Host bridge: Broadcom CMIC-LE 00:04.0 Class ff00: Dell Embedded Remote Access or ERA/O 00:04.1 Class ff00: Dell Remote Access Card III 00:04.2 Class ff00: Dell Embedded Remote Access: BMC/SMIC device 00:0e.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) 00:0f.0 Host bridge: Broadcom CSB5 South Bridge (rev 93) 00:0f.1 IDE interface: Broadcom CSB5 IDE Controller (rev 93) 00:0f.2 USB Controller: Broadcom OSB4/CSB5 OHCI USB Controller (rev 05) 00:0f.3 ISA bridge: Broadcom CSB5 LPC bridge 00:10.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:10.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 01:06.0 Communication controller: Motorola Wildcard X100P 03:06.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 03:08.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 04:08.0 PCI bridge: Intel Corporation 80303 I/O Processor PCI-to-PCI Bridge (rev 01) 04:08.1 RAID bus controller: Dell PowerEdge Expandable RAID Controller 3/Di (rev 01) But, wcfxo won't recognise it: NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 Failed to initailize DAA, giving up... wcfxo: probe of :01:06.0 failed with error -5 This is running on a custom-compiled kernel 2.6.24.3 with Asterisk 1.4.18 and Zaptel 1.4.10. Any ideas? -- Alex Is that a genuine X100P or a clone? Looks like a modem that has been made to look like an X100P. What's the difference? There are some driver peculiarities with some of the Modem-based X100P's that cause them to not be detected correctly by wcfxo. -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. Matt, Thanks for your feedback. We've already tested the following branch as per Shaun's suggestion, without getting a different behaviour (see today's earlier email to the list): http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? We are now in the middle of rebuilding a non 4K stack page kernel so as to give it a try with 1.4.10, the branch Shaun suggested, 1.4.9.2 and the branch you mention, if it is in fact different from Shaun's. We wait your confirmation and will post non 4K stack kernel results later today. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Click-to-call client
equis software wrote: Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? I did a spiel on this at Astricon last year. The slide deck is somewhere around for those interested, but now we also have some code to show for it. :-) Take a look at this developer branch at http://www.asterisk.org/node/48440 and then we've put some pieces together for the Java side of things using Ignite's Realtime API for messaging. http://svn.btwtech.com/svnview/coolvocals/trunk/cti-server/click2call/ http://svn.btwtech.com/svnview/coolvocals/trunk/cti-client/click2call/ Basically the idea here is that there's a servlet that honors requests into it (think AJAX Remote calls from the browser) and then turns around and puts that request into a jabber message that goes to a centralized Servlet that can proxy requests across multiple servers (scalability/LCR/etc) and that in turn launches an Originate call in to the AMI of the machine that was decided would receive the request. Once that hand off is done, the proxy machine that received and directed the original request is now out of the middle of things and jabber messages are sent directly back to the client to signal call progress of the click to call. Is it a shrink wrapped and ready to go package that's completely documented and involves no technical knowledge whatsoever for implementation? U.. no, but that might happen in the relatively near future. :-) What it IS though is solid working code (yes, it has been fully unit tested out and is functional) contributed back to the community so we can all start to make something with it if we so choose. If there's enough interest, I'd certainly entertain opening up a blog site and open up the branch of the Java code for community contributions as well in addition to doing a more detailed tutorial on usage of the code at the upcoming Astricon this year. BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 04:11:52PM +0100, Ex Vito wrote: On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: [snip] Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. Matt, Thanks for your feedback. We've already tested the following branch as per Shaun's suggestion, without getting a different behaviour (see today's earlier email to the list): http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? Try: http://svn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Try the seocnd one (svn.digium.com), actually. All point to the same place. But origsvn does not allow annonymous access and /view is the viewcvs/viewsvn web interface. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. Up till now we're running stock CentOS kernel: 2.6.18-53.1.14.el5 The .config is publicly available but we can fwd it to you should you prefer. The kernel we're now building (it is taking quite a while... but it also has been quite a few years since we've built custom kernels... since the 2.0.3x days ?) is based on the stock CentOS kernel with only the 4K stacks option disabled. Please confirm if the SVN branch you suggested is the same or different from the one Shaun suggested yesterday which we already tested. Thanks, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. Up till now we're running stock CentOS kernel: 2.6.18-53.1.14.el5 The .config is publicly available but we can fwd it to you should you prefer. The kernel we're now building (it is taking quite a while... but it also has been quite a few years since we've built custom kernels... since the 2.0.3x days ?) is based on the stock CentOS kernel with only the 4K stacks option disabled. Please confirm if the SVN branch you suggested is the same or different from the one Shaun suggested yesterday which we already tested. It's the same. Sorry, I sent you that email before I saw his message. I just got an idea for a clever way to make the softlockup detector not complain. I'll let you know when I have a patch to try. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? Try: http://svn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Try the seocnd one (svn.digium.com), actually. All point to the same place. But origsvn does not allow annonymous access and /view is the viewcvs/viewsvn web interface. So Matt's suggestion is the same as Shaun's... Which we already tested with no different results, correct ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 4:46 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: It's the same. Sorry, I sent you that email before I saw his message. I just got an idea for a clever way to make the softlockup detector not complain. I'll let you know when I have a patch to try. ...sure. Thanks. (we're still waiting for the kernel build to finish...) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR and transfers! :(
Well, I think this should be a critical feature to implement for the next releases of Asterisk (hopefully * 1.6), I've read a lot of this matter in the list and in the bug tracker. I think the more insightful reading about this topic can be found in this link: http://www.asterisk.org/node/48358 I know Murf that you are working hard on this, I'm sure all of us on the list encourage you to keep doing your great work... Since the billable time of a call composed of several transfers depends on the policy established for it (by this I mean to whom charge for the whole call), may I suggest that the Future Wonderful CDR needs to include a Policy Manager (the first term that comes to my mind) that allows you to sets how to account for it??? Maybe just a new parameter in cdr*.conf named whotobill or some similar with values that describe the initiator of the call, the person who gets the transfer (in case of a requested call to the operator) and so on... Thanks Greyman for your response, best regards... -- Raul Gomez Linux Counter #156439 On Thu, Apr 17, 2008 at 9:35 AM, Grey Man [EMAIL PROTECTED] wrote: Hi Raul, CDR's for transfers are beyond the ability of Asterisk. http://lists.digium.com/pipermail/asterisk-users/2008-January/204856.html http://bugs.digium.com/view.php?id=11093 It's not something the powers that be want to think about a design for and the solution that's been suggested is to date it to use a different type of server software, such as a SIP Proxy, to generate the CDR's (something easy to suggest and complicated to do). You're not the only one affected by this and there is no fix. Regards, Greyman. -users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Nestor A. Diaz wrote: 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Does a simple sip reload work, or do you really need to go to all the trouble of removing the peer definition? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialed number notify at invalid dial situation
Anonymous wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback(' is incorrect! please again ')
# This exten lines are figure for instruction.
# I know to use with gsm filename.
but ${EXTEN} meaning 'i' that isn't dialed number.
Does anyone have good idea?
please help
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan.
http://www.dairiten.com:81/modules/news/
powered by xoops at http://www.xoops.org
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If you were to use Read to in your IVR instead of Background or
WaitExten, you could then reuse later the variable you read. I haven't
tested this to see if Goto *sends* you to the i extension when you try
to go to a non-existent extension... but *you* could :)
[mainmanu]
exten = s,1,Answer()
exten = s,n,Playback(Press 1, 2, or 3)
exten = s,n,Read(pressedbutton|Press one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
exten = 1,1,blah
exten = 2,1,blah
exten = 3,1,blah
exten = i,1,NoOP(${pressedbutton})
--
*Mojo Wentworth*
HORAN COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
[EMAIL PROTECTED]
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Re: [asterisk-users] Best Click-to-call client
Check the web embedded click-to-call solution from videoreps.net. It is free. It includes click-to-video, click-to-call, and click-to-did CS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Wednesday, April 16, 2008 7:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Best Click-to-call client Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
update with no 4K stack kernel: - The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5 - The only .config change was to disable the CONFIG_4KSTACKS Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as suggested by Shaun and Mathew. Short: Results are about the same (stack traces are different). 1.4.10 and the stackcleanup lead to soft hangups, 1.4.9.2 does not. 1.4.10 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 162 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3613a400 Reg 1: 0x3613a000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels wcte12xp: Clearing yellow alarm BUG: soft lockup detected on CPU#1! [c044d480] softlockup_tick+0x96/0xa4 [c042de00] update_process_times+0x39/0x5c [c04196ef] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [c0605c30] _spin_unlock_irqrestore+0x8/0x9 [f8e82d57] Oct6100UserDriverWriteBurstApi+0x1d/0x27 [wct4xxp] [f8e95de0] Oct6100ApiLoadImage+0x1b5/0x289 [wct4xxp] [f8e9afc4] Oct6100ChipOpen+0x166/0x25e [wct4xxp] [f8e83050] init_vpm450m+0x196/0x306 [wct4xxp] [f8e6ab11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8e6eee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8af6fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f8ac65ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f8acdf6a] zt_ioctl+0x273/0x144f [zaptel] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c0483cb3] __d_lookup+0x98/0xdb [c047b32c] do_lookup+0x53/0x166 [c047d9ec] do_path_lookup+0x20e/0x25e [c0471053] get_empty_filp+0x99/0x15e [c047b5a5] permission+0xa2/0xb5 [c04e1a36] kobject_get+0xf/0x13 [c046ea1e] __dentry_open+0xea/0x1ab [c046eb43] nameidata_to_filp+0x19/0x28 [c046eb7d] do_filp_open+0x2b/0x31 [c047f4a7] do_ioctl+0x47/0x5d [c047f707] vfs_ioctl+0x24a/0x25c [c0470de6] __fput+0x13f/0x167 [c047f761] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 3: Secondary Sync Source Completed startup! 1.4.9.2 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12x[p]: Setting up global serial parameters for T1 wcte12x[p]: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3571b400 Reg 1: 0x3571b000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12x[p]: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 SPAN 2: Primary Sync Source timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 SPAN 3: Secondary Sync Source Completed startup! timing source auto card 0! 1.4-stackcleanup-r4163 dmesg snippet: Zapata Telephony Interface Registered on
[asterisk-users] asterisk trunk
Well, Installed asterisk, libpri, zaptel,... trunk Parameters seems ok for asterisk and ss7, linkset is ok Problem is astersik doesn't matter about the sip messages sent to him, Ngrep see the messages on port 5060 but astersik doesn't react... Even sip set debug on doesn't give me any infos... Any idea someone of what I did wrong? Olivier ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk trunk
Well, Installed asterisk, libpri, zaptel,... trunk Parameters seems ok for asterisk and ss7, linkset is ok Problem is astersik doesn't matter about the sip messages sent to him, Ngrep see the messages on port 5060 but astersik doesn't react... Even sip set debug on doesn't give me any infos... Any idea someone of what I did wrong? Olivier ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PSTN to SIP
Dear all, A quick question on deploying Asterisk over E1. I am looking for a low-cost solution for bridging my E1 line and Asterisk with reasonable stability suppoing both voice and fax. Will a Digium T100 be good for that or I really need a Cisco AS 5400 for this task? What is the difference between using a Digium card vs a physical gateway server? What other alternatives are available? Your suggestions will be greatly appreciated. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Mojo with Horan Company, LLC wrote: Nestor A. Diaz wrote: 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Does a simple sip reload work, or do you really need to go to all the trouble of removing the peer definition? sip reload doesn't work, that's what i have to remove the peer definition, reload, recreate and reload. slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: update with no 4K stack kernel: - The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5 - The only .config change was to disable the CONFIG_4KSTACKS Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as suggested by Shaun and Mathew. Short: Results are about the same (stack traces are different). 1.4.10 and the stackcleanup lead to soft hangups, 1.4.9.2 does not. 1.4.10 dmesg snippet: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Matthew Fredrickson Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 162 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3613a400 Reg 1: 0x3613a000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels wcte12xp: Clearing yellow alarm BUG: soft lockup detected on CPU#1! [c044d480] softlockup_tick+0x96/0xa4 [c042de00] update_process_times+0x39/0x5c [c04196ef] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [c0605c30] _spin_unlock_irqrestore+0x8/0x9 [f8e82d57] Oct6100UserDriverWriteBurstApi+0x1d/0x27 [wct4xxp] [f8e95de0] Oct6100ApiLoadImage+0x1b5/0x289 [wct4xxp] [f8e9afc4] Oct6100ChipOpen+0x166/0x25e [wct4xxp] [f8e83050] init_vpm450m+0x196/0x306 [wct4xxp] [f8e6ab11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8e6eee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8af6fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f8ac65ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f8acdf6a] zt_ioctl+0x273/0x144f [zaptel] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c0483cb3] __d_lookup+0x98/0xdb [c047b32c] do_lookup+0x53/0x166 [c047d9ec] do_path_lookup+0x20e/0x25e [c0471053] get_empty_filp+0x99/0x15e [c047b5a5] permission+0xa2/0xb5 [c04e1a36] kobject_get+0xf/0x13 [c046ea1e] __dentry_open+0xea/0x1ab [c046eb43] nameidata_to_filp+0x19/0x28 [c046eb7d] do_filp_open+0x2b/0x31 [c047f4a7] do_ioctl+0x47/0x5d [c047f707] vfs_ioctl+0x24a/0x25c [c0470de6] __fput+0x13f/0x167 [c047f761] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 3: Secondary Sync Source Completed startup! 1.4.9.2 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12x[p]: Setting up global serial parameters for T1 wcte12x[p]: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3571b400 Reg 1: 0x3571b000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12x[p]: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 SPAN 2: Primary Sync Source timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels VPM450: hardware DTMF disabled.
[asterisk-users] Chanspy on Asterisk 1.4.19
Hi all,
I've just upgraded to 1.4.19 from 1.4.18.1 and now have problems with
app_chanspy. To monitor I use -
exten = 596,1,ringing
exten = 596,n,Wait(1)
exten = 596,n,ChanSpy(|g(2000))
exten = 596,n,Hangup
and the listened-to channel as follows -
exten = _77,1,Set(SPYGROUP=2000)
exten = _77,n,Dial(Zap/g2/${EXTEN:2})
This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered
but there's no spying, the only way I could get this to work was with -
exten = 596,n,ChanSpy(|b)
but this spied on all channels, not just those with SPYGROUP set to 2000
so not much use to us.
I've recompiled Asterisk 1.4.19 with app_chanspy.c from 1.4.18.1 and it
works again. I'm using latest zaptel, libpri and addons on CentOS 4.4.
Changelog in 1.4.19 shows some changes to app_chanspy to stop asterisk
crashes and other improvements so would be nice to have the fixes maybe.
Anyone any ideas?
Regards,
Steve
___
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[asterisk-users] Using Chanspy
Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set SPYGROUP) line. Well, it's hit and miss. Sometimes it does, sometimes it doesn't, and sometimes it even kill me original communication. What am I missing? Or is Chanspy not working as designed? Using Asterisk 1.4.19. Regards, Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN to SIP
If your requirements are simple and you only have a small number if E1s, you can also use a Cisco 36xx with a T1/PRI card. 3600's have limited capacity but we run 4 PRIs on a 3640 no problem and it's been very stable for several years. The nice thing about 3600's is they are almost free, although the cards are not. Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 17 Apr 2008, mark morreny wrote: Dear all, A quick question on deploying Asterisk over E1. I am looking for a low-cost solution for bridging my E1 line and Asterisk with reasonable stability suppoing both voice and fax. Will a Digium T100 be good for that or I really need a Cisco AS 5400 for this task? What is the difference between using a Digium card vs a physical gateway server? What other alternatives are available? Your suggestions will be greatly appreciated. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compilation of asterisk 1.4.19 with ilbc already on system
--- Kevin P. Fleming [EMAIL PROTECTED] wrote: Vieri wrote: How can I tell the make system in 1.4.19 that ilbc is already on the system and that it should link to /usr/lib/libilbc.a? Shouldn't the configure script do that? No; the Asterisk build system has never had support for using a system-provided version of the iLBC library. Whoever provided you that library could easily run afoul of the same licensing issues that caused us to remove the code from our Asterisk distribution, and using that library does not obviate you from the need to register your intent to use the codec if you are using it for commercial purposes. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. Thanks but I'm curious. I'm using Gentoo Linux and there's an ebuild (ie. package) to install the iLBC development library. Gentoo ebuilds are scripts that automatically download upstream source code, compile and install (no binary packages). As far as licensing is concerned, all ebuilds are required to specify the type of license. So in case of iLBC, the LICENSE keyword points to http://www.ilbcfreeware.org/documentation/gips_iLBClicense.pdf. Thus, when iLBC is installed via this ebuild the user knows the license its under (and therefore accepts it whether for commercial or personal use). I don't see the legal problem with installing iLBC this way. So I had to slightly change the asterisk build process: 1) modify the codecs Makefile: --- codecs/Makefile.orig2008-04-14 12:48:09.0 +0200 +++ codecs/Makefile 2008-04-14 12:49:46.0 +0200 @@ -29,7 +29,7 @@ LOADABLE_MODS:= endif -LIBILBC:=ilbc/libilbc.a +LIBILBC:=/usr/lib/libilbc.a LIBLPC10:=lpc10/liblpc10.a all: _all @@ -56,6 +56,6 @@ $(if $(filter codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10) $(LIBILBC): - @$(MAKE) -C ilbc all ASTCFLAGS=$(filter-out -Wmissing-prototypes -Wmissing-declarations,$(ASTCFLAGS)) $(AST_NO_STRICT_OVERFLOW) + @echo Using /usr/lib/libilbc.a $(if $(filter codec_ilbc,$(EMBEDDED_MODS)),modules.link,codec_ilbc.so): $(LIBILBC) 2) remove codec_ilbc from MENUSELECT_CODECS in menuselect.makeopts so that codec_ilbc gets built by asterisk: make menuselect.makeopts sed -i -e s:codec_ilbc:: menuselect.makeopts 3) make asterisk as usual So basically I'm wondering if the Asterisk make/configure process could do steps 1 and 2 automagically for me. Vieri Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... Can you answer my previous questions ? - If going live would you recommend zaptel 1.4.9.2 or 1.4.10 ? - Does the current behaviour from 1.4.10 prevent firmware uploading ? (or, stated differently: can you explain what is happening that makes the system hang for a few seconds ?) Thanks, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Chanspy
Try using chanspy without setting the variable first. This should give you a broader base of channels. Then start to narrow it down. On Wed, Apr 16, 2008 at 8:33 PM, Mike [EMAIL PROTECTED] wrote: Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set SPYGROUP) line. Well, it's hit and miss. Sometimes it does, sometimes it doesn't, and sometimes it even kill me original communication. What am I missing? Or is Chanspy not working as designed? Using Asterisk 1.4.19. Regards, Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Can you answer my previous questions ? - If going live would you recommend zaptel 1.4.9.2 or 1.4.10 ? I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. - Does the current behaviour from 1.4.10 prevent firmware uploading ? No. There is nothing that is happening that prevents firmware uploading. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with hints (1.4.19)
Hi, (me again, my upgrade to 1.4 is more painful then I imagined it would be). I just noticed that the command show hints shows all hints correctly, but none of them ever are InUse (even if I use a line and dial out) like I used to on 1.2. Can`t find a bug in the bug tracking system, is there something else I should be doing in 1.4.19 for it to work? Thanks, Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Drag and Drop transfer application
Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
--- Nestor A. Diaz [EMAIL PROTECTED] wrote: Mojo with Horan Company, LLC wrote: Nestor A. Diaz wrote: 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Did you try a show channels to see if there were stale channels for peer 200? I had the same problem you describe but it was due to hung channels (used * 1.4.18.1 with rtp*timeout and saw inuse peers during the pre-timeout periods even though the agents weren't on a call). Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialed number notify at invalid dial situation
Mojo with Horan Company, LLC wrote:
[mainmanu]
exten = s,1,Answer()
exten = s,n,Playback(Press 1, 2, or 3)
exten = s,n,Read(pressedbutton|Press one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
Oops,
shouldn't have that second priority in there. Because Read is playing
the prompt, Playback is unnecessary.
Moj
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Re: [asterisk-users] Using Chanspy
Mike wrote: Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) I know it's unlikely, but could some of the dialplan changes from 1.6 have accidentally filtered backwards into the 1.4 tree? ie. Chanspy(|qg(1234)) becomes Chanspy(,qg(1234)) Unlikely I know, but probably worth a shot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Chanspy
|=, as of 1.2 IIRC On Wed, Apr 16, 2008 at 9:54 PM, Thomas Kenyon [EMAIL PROTECTED] wrote: Mike wrote: Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) I know it's unlikely, but could some of the dialplan changes from 1.6 have accidentally filtered backwards into the 1.4 tree? ie. Chanspy(|qg(1234)) becomes Chanspy(,qg(1234)) Unlikely I know, but probably worth a shot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_zap error 1.4.19 tone duration
I am getting an error: chan_zap invalid tone duration 11220. This is line 11220 in chan_zap.c and I have a toneduration of 300 in the zapata.conf file. I have commented it out and it is now working again. Why is that an invalid paramter? It never used to be. jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Al lists wrote: Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Flash Operator Panel (FOP) http://www.asternic.org -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_zap error 1.4.19 tone duration
On Wed, Apr 16, 2008 at 03:51:02PM -0400, Jerry Geis wrote: I am getting an error: chan_zap invalid tone duration 11220. This is line 11220 in chan_zap.c and I have a toneduration of 300 in the zapata.conf file. I have commented it out and it is now working again. Why is that an invalid paramter? It never used to be. http://bugs.digium.com/12456 (Right now there's no useful information there for you, though) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Al lists wrote: Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? Yes. Maestro Control Panel (I authored this one) http://www.datatrakpos.com/pos/datatalk/maestro.aspx. There is also the nice flash based Flash Operator Panel http://www.datatrakpos.com/pos/datatalk/maestro.aspx There a couple of other ones out there too that I thought were nice, but can't remember the names. You should be able to find them by gooling for Asterisk Control Panel or such query. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Introducing Click-to-Call Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. The 1Ezphone web-based Click-to-Call service is based on our browser VoIP lite technology that allows users to make and receive phone calls from any browser without the need to download software. The Click-to Call API can be embedded on any Website, E-mail, and Online Advertisement when a user clicks your object they immediately call your salesperson or customer service representative telephone number and speak to your agent over their PC. Building a reliable Click-to-Call requires substantial amount of knowledge in VoIP, and a good backend infrastructure. The good news is that now it is easy add Click-to Call to any online service in just a few minutes with just a few lines of code using 1ezphones. Since the release of our APIs, we got several requests from companies and developers who were interested in knowing in building their own Click-to-Call service directly to their SIP servers. You can have the button/widget running through the 1Ezphones servers without getting into the complex world of VoIP or any expensive setup or build a service to your own backend infrastructure. If you are interested in adding Click-to-Call for your customers or building your own Click-to Call system please contact 1ezphone at [EMAIL PROTECTED] - Original Message - From: Al lists To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Drag and Drop transfer application Date: Wed, 16 Apr 2008 12:27:12 -0600 Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
On Wed, 2008-04-16 at 18:51 +0100, Steve Rawlings wrote: This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered but there's no spying, the only way I could get this to work was with - exten = 596,n,ChanSpy(|b) but this spied on all channels, not just those with SPYGROUP set to 2000 so not much use to us. I've recompiled Asterisk 1.4.19 with app_chanspy.c from 1.4.18.1 and it works again. I'm using latest zaptel, libpri and addons on CentOS 4.4. Changelog in 1.4.19 shows some changes to app_chanspy to stop asterisk crashes and other improvements so would be nice to have the fixes maybe. It sounds like there may have been some sort of regression introduced in the changes between 1.4.18.1 and 1.4.19. Would you mind opening a ticket on the bug tracker so that the developers can make sure it gets addressed? -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with hints (1.4.19)
On Wed, 2008-04-16 at 14:20 -0400, Mike wrote: I just noticed that the command show hints shows all hints correctly, but none of them ever are InUse (even if I use a line and dial out) like I used to on 1.2. Can`t find a bug in the bug tracking system, is there something else I should be doing in 1.4.19 for it to work? For SIP devices, you need to have the call-limit setting set so that Asterisk will keep track of the device state. I typically set call-limit=99 on the devices for which I've built hints. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN to SIP
Rhino or audiocode PSTN gateway - Original Message - From: mark morreny To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PSTN to SIP Date: Thu, 17 Apr 2008 01:25:49 +0800 Dear all, A quick question on deploying Asterisk over E1. I am looking for a low-cost solution for bridging my E1 line and Asterisk with reasonable stability suppoing both voice and fax. Will a Digium T100 be good for that or I really need a Cisco AS 5400 for this task? What is the difference between using a Digium card vs a physical gateway server? What other alternatives are available? Your suggestions will be greatly appreciated. Thanks,Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Click-to-call client
Introducing Click-to-Call Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. The 1Ezphone web-based Click-to-Call service is based on our browser VoIP lite technology that allows users to make and receive phone calls from any browser without the need to download software. The Click-to Call API can be embedded on any Website, E-mail, and Online Advertisement when a user clicks your object they immediately call your salesperson or customer service representative telephone number and speak to your agent over their PC. Building a reliable Click-to-Call requires substantial amount of knowledge in VoIP, and a good backend infrastructure. The good news is that now it is easy add Click-to Call to any online service in just a few minutes with just a few lines of code using 1ezphones. Since the release of our APIs, we got several requests from companies and developers who were interested in knowing in building their own Click-to-Call service directly to their SIP servers. You can have the button/widget running through the 1Ezphones servers without getting into the complex world of VoIP or any expensive setup or build a service to your own backend infrastructure. If you are interested in adding Click-to-Call for your customers or building your own Click-to Call system please contact 1ezphone at [EMAIL PROTECTED] - Original Message - From: BJ Weschke To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Click-to-call client Date: Wed, 16 Apr 2008 11:37:40 -0400 equis software wrote: Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? I did a spiel on this at Astricon last year. The slide deck is somewhere around for those interested, but now we also have some code to show for it. :-) Take a look at this developer branch at http://www.asterisk.org/node/48440 and then we've put some pieces together for the Java side of things using Ignite's Realtime API for messaging. http://svn.btwtech.com/svnview/coolvocals/trunk/cti-server/click2call/ http://svn.btwtech.com/svnview/coolvocals/trunk/cti-client/click2call/ Basically the idea here is that there's a servlet that honors requests into it (think AJAX Remote calls from the browser) and then turns around and puts that request into a jabber message that goes to a centralized Servlet that can proxy requests across multiple servers (scalability/LCR/etc) and that in turn launches an Originate call in to the AMI of the machine that was decided would receive the request. Once that hand off is done, the proxy machine that received and directed the original request is now out of the middle of things and jabber messages are sent directly back to the client to signal call progress of the click to call. Is it a shrink wrapped and ready to go package that's completely documented and involves no technical knowledge whatsoever for implementation? U.. no, but that might happen in the relatively near future. :-) What it IS though is solid working code (yes, it has been fully unit tested out and is functional) contributed back to the community so we can all start to make something with it if we so choose. If there's enough interest, I'd certainly entertain opening up a blog site and open up the branch of the Java code for community contributions as well in addition to doing a more detailed tutorial on usage of the code at the upcoming Astricon this year. BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Lee Jenkins wrote: Al lists wrote: Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? Yes. Maestro Control Panel (I authored this one) http://www.datatrakpos.com/pos/datatalk/maestro.aspx. There is also the nice flash based Flash Operator Panel http://www.datatrakpos.com/pos/datatalk/maestro.aspx Oops. Sorry, for FOP that is: http://www.asternic.org/ NoteToSelf note=Stop replaying to email while on the phone/ -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
On Wed, Apr 16, 2008 at 9:10 AM, broadband Voice [EMAIL PROTECTED] wrote: We have two servers but looks like G729 issues. Works fine on the old server and not sure if it is T1 related. See SIP Debug. Any experiences to share. Thanks --- Newark1*CLI --- SIP read from 194.xx.Xx.Xx:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=K784d2637;rport From: Cell Phone DC sip:[EMAIL PROTECTED];tag=as04819ca3 To: sip:xx;tag=xx Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 198 v=0 o=xx 12x 12 IN IP4 62.xx.xx.xx s=SIP Call c=IN IP4 62.xx.xx.xxx t=0 0 m=audio 8786 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 - --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 62.xx.xx.xx:8786 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 62.xx.xx.xx:8786 -- SIP/Voicetrading-08e1ce18 is making progress passing it to Zap/5-1 Looks to be OK to me but you have negotiated Ulaw not G729. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Bob G wrote: Introducing Click-to-Call http://1ezphone.com/ Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. I think you're going to get yelled at ;) -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extenspy and chanspy
I want to add to my dialplan the ability to spy on an arbitrary extension whether a call originates at it or is terminated at it. Scenario 1: Given an extension, say 2001, a call comes in on a zap channel and is Dial()ed to the phone that's at extension 2001, I want to be able to pick up a phone and dial (say) *142001 and spy on that call. Scenario 2: Extension 2001 makes a call to, say a zap channel, again, I want to be able to pick up a phone and dial *142001 and spy on that call. ExtenSpy(exten@context) seems like the obvious first choice but it requires a context and an extension. I think that can work for scenario 2 but not scenario 1, yes? An extension answering a call doesn't have a context does it? Alternatively I could use SPYGROUP and assign a SPYGROUP when an outbound call is made but I think I would need some way for Dial to set a SPYGROUP when an extension answers wouldn't I? Does anyone have an implementation of this they'd like to share? b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Why the guy asked a question? - Original Message - From: Lee Jenkins To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Drag and Drop transfer application Date: Wed, 16 Apr 2008 16:21:54 -0400 Bob G wrote: Introducing Click-to-Call Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. I think you're going to get yelled at ;) -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compilation of asterisk 1.4.19 with ilbc already on system
Vieri wrote: So basically I'm wondering if the Asterisk make/configure process could do steps 1 and 2 automagically for me. I can't find any other Linux distribution that provides libilbc, so this would be a very Gentoo-specific change if we did it. Also, we'll have the iLBC source code back in the main distribution in the near future when the licensing issues are worked out, so for everyone else this will become a non-issue. Do you see any particular advantage to using the system-provided libilbc, given that we use it in static (not shared object) form and it would have to be relinked into Asterisk if it got upgraded anyway? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
On Wed, 16 Apr 2008, Bob G wrote: Introducing Click-to-Call So, since you posted this on a non-commercial discussion list, this is available for free? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
On Wed, 16 Apr 2008, Bob G wrote: Why the guy asked a question? From: Lee Jenkins Bob G wrote: Introducing Click-to-Call I think you're going to get yelled at ;) 1) You hijacked the thread. 2) You top-posted. 3) It's a non-commercial list -- RTFMLIBP (Read the Mailing List Instructions Before Posting). Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extenspy and chanspy
Brian J. Murrell wrote:
Does anyone have an implementation of this they'd like to share?
I cut out the authentication stuff we do, but this is part of the macro
we use to spy and record calls arbitrary calls. All of our users have
sip handsets. Asterisk 1.2.
exten = s,n(getext),Read(SPY,extension,4)
exten = s,n,GotoIf($[ ${LEN(${SPY})} != 4 ]?nospy)
exten = s,n(spy),UserEvent(ChanSpy,User ${CALLBACKNUM} spied on ${SPY})
exten = s,n,Chanspy(SIP/${SPY},r(monitor-ext-${SPY}))
exten = s,n,Hangup()
exten = s,n(nospy),Playback(sorry-cant-let-you-do-that3)
exten = s,n,UserEvent(ChanSpy,User ${CALLBACKNUM} failed to spy on ${SPY})
exten = s,n,Hangup()
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Re: [asterisk-users] chan_zap error 1.4.19 tone duration
Jerry Geis wrote: I am getting an error: chan_zap invalid tone duration 11220. Is this actually the error message you got? I don't see the line number being placed into the error message by the code in chan_zap. When reporting errors, it is very helpful if you actually copy and paste the error message instead of typing it in 'from memory'. This is line 11220 in chan_zap.c and I have a toneduration of 300 in the zapata.conf file. I have commented it out and it is now working again. It's not working, because your tone duration is not actually being set. This is a bug in Asterisk, which has been corrected in Subversion branch 1.4 and will be in the next 1.4 release. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Steve Edwards wrote: On Wed, 16 Apr 2008, Bob G wrote: Why the guy asked a question? From: Lee Jenkins Bob G wrote: Introducing Click-to-Call I think you're going to get yelled at ;) 1) You hijacked the thread. 2) You top-posted. 3) It's a non-commercial list -- RTFMLIBP (Read the Mailing List Instructions Before Posting). 4) You just used a run-on sentence. ;) -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Bob G wrote: Why the guy asked a question? Yes. But the question was about Drag and Drop transfer applications for Asterisk. Can 1EZphone do that? If not, your SPAMMING the list! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
On Wed, Apr 16, 2008 at 03:24:15PM -0500, Bob G wrote: Why the guy asked a question? And you did not provide a useful answer to it. You merely quoted a leangthy press release. It might have been partially relevant. And might not. Stick to relevant answers. Certainly so when promoting your commercial products in this non-commercial list. As a rule: don't promote your product in this list. If it is that good, your more impartial users will recommend it for you. I realize that this is not completely practical: I work for a certain company and know our products well. I know that they fit well for certain things. And anyway, if you have a hammer in your hand, everything is a nail. But do realise that your extra promotion paints you as unrliable. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extenspy and chanspy
On Wed, 2008-04-16 at 13:47 -0700, Steven Kurylo wrote:
exten = s,n(getext),Read(SPY,extension,4)
exten = s,n,GotoIf($[ ${LEN(${SPY})} != 4 ]?nospy)
exten = s,n(spy),UserEvent(ChanSpy,User ${CALLBACKNUM} spied on ${SPY})
exten = s,n,Chanspy(SIP/${SPY},r(monitor-ext-${SPY}))
exten = s,n,Hangup()
exten = s,n(nospy),Playback(sorry-cant-let-you-do-that3)
exten = s,n,UserEvent(ChanSpy,User ${CALLBACKNUM} failed to spy on ${SPY})
exten = s,n,Hangup()
Ahhh. And this works because your sip.conf entries are of the form:
[exten_num]
...
I see. I name mine more symbolically, but I wonder if I could create
some kind of mapping... or take advantage of the callerid attribution
of a sip.conf entry somehow to map from $callerid_number to $entry_name.
Cheers!
b.
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[asterisk-users] QOS for outgoing SIP calls
Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Sorry - Original Message - From: Tzafrir Cohen To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Drag and Drop transfer application Date: Thu, 17 Apr 2008 00:00:57 +0300 On Wed, Apr 16, 2008 at 03:24:15PM -0500, Bob G wrote: Why the guy asked a question? And you did not provide a useful answer to it. You merely quoted a leangthy press release. It might have been partially relevant. And might not. Stick to relevant answers. Certainly so when promoting your commercial products in this non-commercial list. As a rule: don't promote your product in this list. If it is that good, your more impartial users will recommend it for you. I realize that this is not completely practical: I work for a certain company and know our products well. I know that they fit well for certain things. And anyway, if you have a hammer in your hand, everything is a nail. But do realise that your extra promotion paints you as unrliable. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lightweight prepaid app using Dial and extentions.conf
I have just noticed the L() argument to Dial() and it seems pretty obvious that this could be used to create a lightweight prepaid calling system. I'm wondering if anyone has some extensions.conf dialplan using Dial(..., L(...)) and the astdb to do lightweight prepaid service. I only need to meter a handful of users. Cheers, b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and LVS
Has anyone used or thought of using Asterisk server farm behind LVS. -Jai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup conundrum with RxFAX
Gordon Henderson wrote: Heres something that's making me scratch my head... I'm using RxFAX on ISDN lines and in-general it's going well. However, there seems to be a case when the fax doesn't get delivered, but looking through the CDRs it seems that the call happened, RxFAX was executed .. time passed (1-2+ minutes) then hangup. I'm wondering if some FAX machines just hangup after the call rather than complete some sort of ending negotiation, or if the RxFAX part misses the end and just sees the hangup.. Now, in a normal fax machine, it's going to print the fax regardless, even if the last page is only half full because of a genuine line drop or hangup, but it seems that: [Description] RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the ... Returns -1 when the user hangs up. Returns 0 otherwise. So if it's returning -1, then the call/channel is hungup, and any dialplan instructions after it won't get executed, even though there might be some (or all) pages of the fax sitting in the receive file... Does this make sense to anyone, or am I barking up the wrong tree! My thoughts now are to actually do a hangup at the end of the RxFAX and rely on a 'h' extension to pick it up and carry on with the 2nd half (which is PDFing and emailling the fax), but I'm concerned I'm going to lose the channel variables as it suggests on the wiki, so I'll lose the REMOTESTATIONID string and caller ID... Anyone with any experience of this, or suggestions otherwise? Thanks, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi. Thats what I do and have not had a problem, we only do maybe 10-20 faxes a week though. I set my channel variables in a macro and then goto a context receivefax where I enter on s,1,Rx.Fax , on hangup I do the actual mailing and sending of the fax. Before the sending though I make sure the fax actually exists. hth Jeremy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PSTN to SIP
Hi, Our requirement is just to be able to do voice and fax at a quality manner. What is the difference between using a physical server vs a PCI card that plugs in to the Asterisk server? Is there a big difference in terms of scalability? We are looking at a solution that can be easy-to-deploy ourselves and reasonable voice and fax quality. Thanks for your inputs. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
On Wed, Apr 16, 2008 at 11:49 PM, Simon [EMAIL PROTECTED] wrote: Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Hi Simon, You won't be able to get much use of your router's QoS if it can only set it via port number. By default Asterisk will select a UDP port somewhere in the range of 10,000 to 20,000 to carry the RTP. The port selected for the RTP will be different at your end and at your providers end which means you would need two QoS port rules per call. You can change the port range your Asterisk server uses for RTP in rtp.conf but there's probably not a lot of point given you can't prioritise a big enough range with only 3 rules available. To be of any practical use for SIP calls you really need to be able to set QoS by IP address. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN to SIP
The Cisco's also support T.38 gateway functions whereas Asterisk can only do pass thru. Either way you'll still need another server, typically hylafax, to receive the faxes to get them somewhere useful. In my experience the Cisco switches are definitely the way to go for the ISDN/SIP gateway and then back it on to Asterisk for any PBX bells and whistles needed. If you've got more time and patience than money then a card in Asterisk for the ISDN gateway is an ok solution. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] keep one line open
hi i have multiple lines going to my asterisk box etc 0282549087 , 028 3659874 , 0285469658 etc. is it possible to keep users from using the 0282549087 line always open that it only allows a certain user to make outgoing calls on it? - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
