[Asterisk-Users] Re: asterisk with german SIPGATE ?
Hi, whats is your problem ? For me it works, but had problems too. nicolas Thorsten Gehrig wrote: hi anybody running with german SIPGATE? my configuration don't works :-( regards [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] x100p / Answer- Flash - Dial
Title: Message Dan, This will probably not work. Once Asterisk tells the Panasonic PBX to transfer the call, the call will no longer go through the extension the Asterisk PBX is attached to. It seems the only solution to this (doing it the way you are) is to have two zaptel cards in the Asterisk PBX, attache them to two extensions of the Panasonic PBX and the calls come in one extension and then are sent out the other KEEPING BOTH BUSY as long as the call is in progress. As Sam said, you will really need to have an intelligent connection to the Panasonic PBX to do this on much more than a single call at a time basis. Disclaimer: I am an Asterisk Newbie, though I have been following it for several years, my hands on experience is abouta week old. Andy Farnsworth -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam BingnerSent: Saturday, May 08, 2004 10:54 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] x100p / Answer- Flash - Dial Even if you could get that to work properly, which I dont know... the callprogress detection is horrible; if you want to do that reliably you need a T1,ISDN or IPinterface to the switch (something that actually provides proper call progress) Sam -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan FernandezSent: Saturday, May 08, 2004 11:44 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] x100p / Answer- Flash - Dial I have an X100P connected to an extension of aPanasonic PBX.When a call from the PSTN comes in,it is routed directly to theextension where the x100p is .I want* to answer the call, play amessage and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then todialan IAX channel. The problem is that when I do a Flash on thezap channel, and then try to dial a new extensionvia that zap channel I get the following error "can't createzap channel". If I do a SendDTMF()thecalldoes get transfer to the new extension but then * gets out of the callloop and don't know it is answered or not by the new extension. AmI missing something? Why am I getting the "can't creatza channel" Thanksin advance. Dan
[Asterisk-Users] asterisk/can_capi took ISDN B Channels busy.
Hi, i want use both B channels on my isdn card (B1 ISA) but chan_capi open one channel and asterisk say 2. channel is busy. Must i use another isdn card ? I have a old B1 ISA card. Can anyone help me with that ? nicolas SNIPS: -- == DISCONNECT_IND PLCI=0x201 REASON=0x34a2 -- CAPI[contr1/outgoingmsn]/78 is busy -- CAPI Hangingup -- removed pipe for PLCI = 0x201 == Everyone is busy at this time -- Executing Congestion(Local/nummer to forward@default-f1df,2, ) in new stack -- Local/nummer to forward@default-f1df,1 is circuit-busy == Spawn extension (default, nummer to forward, 2) exited non-zero on 'Local/nummer to forward@default-f1df,2' == Everyone is busy at this time -- Executing DigitTimeout(CAPI[contr1/outgoingmsn]/77, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(CAPI[contr1/outgoingmsn]/77, 10) in new stack -- Set Response Timeout to 10 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telekom ISDN CFU is it possible ?
Hello i have the question: is it possible to make a CFU like a isdn phone at the telekom it do ? nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that you keep using, this is a
reminder to check into the configs/ directory of the Asterisk source tree, regardless
if you downloaded a tar ball or from CVS.
Good advice - so I do a CVS UPDATE... and 'say.c' is broken
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-04/05/04-09:58:21\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o say.o say.c
say.c: In function `ast_say_digit_str':
say.c:50: syntax error before '' token
say.c:57: warning: no return statement in function returning non-void
say.c: At top level:
say.c:58: syntax error before if
and in 'say.c' at about line 50
case ('#'):
snprintf(fn, sizeof(fn),
/digits/pound);
break;
default:
say.c
snprintf(fn, sizeof(fn), /digits/%c,
fn2[num]);
}
===
if((fn2[num] = '0') (fn2[num] =
'9')){ /* Must be in {0-9} */
snprintf(fn, sizeof(fn),
digits/%c, fn2[num]);
}
--
The lines that begin with say.c
-or is this just an error caused by CVS
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
On Sun, 2004-05-09 at 14:33, Mark Elkins wrote: On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote: * Read the config sample files! (even if you're an Asterisk guru) - For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. Good advice - so I do a CVS UPDATE... and 'say.c' is broken ... The lines that begin with say.c Sorry folks... seems like a CVS Update did break - removed the file and re-updated. fine now. However - this could bit other people too.. in which case - delete the offending file - and update again (or always use 'cvs checkout' - less efficient - but..) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Where to start?
Hello, I manage a small office and we have a 4-year old legacy analog PBX manufactured by Iwatsu. We have four incoming analog lines that terminate to 7 different desktop phones. The interface to Iwatsu requires Windows and the Iwatsu admin tools are proprietary and not extensible. What do I need to do to begin using Asterisk and/or Digium to create an open source PBX situation? I do not know where to begin in terms of harware needed, changes to service levels from my telco (Sprint), etc. Can I leverage any of my existing PBX technology? Thank you in advance, Ed - Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book http://support.ucompass.com/addressbook.html Committed to Building Profitable E-Learning Enterprises Phone: (850) 297 1800 x 201 FAX: (850) 553-9252 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy problem?!?
Hi, ztdummy says the following: VoiceBOX:/usr/src/zaptel# modprobe ztdummy /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_register /lib/modules/2.4.18/misc/ztdummy.o: insmod /lib/modules/2.4.18/misc/ztdummy.o failed /lib/modules/2.4.18/misc/ztdummy.o: insmod ztdummy failed I have the uhci_usb modules etc. installed. zaptelrtc says nearly the same (./zaprtc.o: unresolved symbol zt_transmit). What's the problem here? Regards, Thoms ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where to start?
Ed, Are you keeping the desktop phones, or upgrading to new phones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Mansouri Sent: 09 May 2004 14:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Where to start? Hello, I manage a small office and we have a 4-year old legacy analog PBX manufactured by Iwatsu. We have four incoming analog lines that terminate to 7 different desktop phones. The interface to Iwatsu requires Windows and the Iwatsu admin tools are proprietary and not extensible. What do I need to do to begin using Asterisk and/or Digium to create an open source PBX situation? I do not know where to begin in terms of harware needed, changes to service levels from my telco (Sprint), etc. Can I leverage any of my existing PBX technology? Thank you in advance, Ed - Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book http://support.ucompass.com/addressbook.html Committed to Building Profitable E-Learning Enterprises Phone: (850) 297 1800 x 201 FAX: (850) 553-9252 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where to start?
Hello, The goal would be to minimize the expenditure on new equipment, and utilize as much of the existing equipment as possible, so I guess it just depends on what the requirements of working with an open source PBX are. The current desktop phones are pretty sophisticated and provide a lot of functionality the preference would be to continue to employ them. Thank you in advance for your insight. Ed - Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book http://support.ucompass.com/addressbook.html Committed to Building Profitable E-Learning Enterprises Phone: (850) 297 1800 x 201 FAX: (850) 553-9252 On Sun, 9 May 2004 [EMAIL PROTECTED] wrote: Ed, Are you keeping the desktop phones, or upgrading to new phones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Mansouri Sent: 09 May 2004 14:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Where to start? Hello, I manage a small office and we have a 4-year old legacy analog PBX manufactured by Iwatsu. We have four incoming analog lines that terminate to 7 different desktop phones. The interface to Iwatsu requires Windows and the Iwatsu admin tools are proprietary and not extensible. What do I need to do to begin using Asterisk and/or Digium to create an open source PBX situation? I do not know where to begin in terms of harware needed, changes to service levels from my telco (Sprint), etc. Can I leverage any of my existing PBX technology? Thank you in advance, Ed - Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book http://support.ucompass.com/addressbook.html Committed to Building Profitable E-Learning Enterprises Phone: (850) 297 1800 x 201 FAX: (850) 553-9252 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 500ms usleep in rtp.c ?
http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(50); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. Can't say that I've noticed any at all, but then most of our calls are tdm04b-fxo -- 7960's with very little bridging. Since I recall you mentioning your use of 7960's previously, could the blip be related to the cisco issue/problem associated with slow startup and/or their v6.x code that drops rtp packets with inconsistence timestamps, combined with the usleep parameter? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 500ms usleep in rtp.c ?
Nope this was from my sipura ... bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, May 09, 2004 9:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 500ms usleep in rtp.c ? http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(50); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. Can't say that I've noticed any at all, but then most of our calls are tdm04b-fxo -- 7960's with very little bridging. Since I recall you mentioning your use of 7960's previously, could the blip be related to the cisco issue/problem associated with slow startup and/or their v6.x code that drops rtp packets with inconsistence timestamps, combined with the usleep parameter? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip to PSTN Gateway Configs
Hi, I'm trying to put together a simple gateway configuration involving Asterisk. I have a machine with 2 Digium X100P FXO cards installed and the Asterisk Software, and I have 2 Sip Phones defined. What I want to achieve is, any call arriving at FXO 1 is forwarded to Sip phone 1 only, and any call received on FXO 2 is transferred to Sip Phone 2, conversely any call originating from Sip Phone 1 goes out of FXO 1, and any call originating from Sip Phone 2 goes out of FXO 2. Some example .conf files would be greatly appreciated. Thanks Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Re: asterisk with german SIPGATE ?
Hi experts (hope so), I´ve behind a firewall (Linux FLI4L) - but i have configured all possible Forwardings. Two Problems at this time: a) after many tries I have registered on siptel *CLI sip show registry Host Username Refresh State 217.10.79.9:5060 8003440 120 Registered I can make calls - but I can´t hear anything! Asterisk shows: -- Executing Dial(SIP/thorstengehrig-2641, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- SIP/sipgate1-dd8a answered SIP/thorstengehrig-2641 -- Attempting native bridge of SIP/thorstengehrig-2641 and SIP/sipgate1-dd8a I think the problem is that the RDP is not coming to the Asterisk? SIPGate-Website shows me as online! b) the second Problem is that Phonecalls to Sipgate are not forwarded to my Asterisk (I cant see anything on the console). (but i´m in Registered state and Website shows me online). I´ve configurated the sip-parts with qualify=yes and I recive this information: *CLI sip show peers Name/usernameHost Mask Port Status thorstengehrig/ 192.168.0.100 (D) 255.255.255.255 5060 OK (2 ms) sipgate1/800344 217.10.79.9 255.255.255.255 5060 OK (118 ms) My Router is 0n 192.168.0.1, my Asterisk Server is 192.168.0.105 Here are my portforwardings: PORTFW_10='5004 192.168.0.105 UDP' # Port für SIP (RTP) zum Debian PORTFW_11='5060 192.168.0.105 tcp' # Port für SIP TCP zum Debian PORTFW_12='5060 192.168.0.105 udp' # Port für SIP UDP zum Debian PORTFW_13='5060 192.168.0.105 tcp' # Port für SIP UDP zum Debian PORTFW_14='5070-5080 192.168.0.105 udp' # für das RTP vom SIP-Phone PORTFW_24='8000-8012 192.168.0.105 udp' # SIPGATE ?!?! PORTFW_25='1-2 192.168.0.105 udp'# SIP ?!?! here is my SIP.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls srvlookup = no ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel tos=lowdelay; Type of Service kiwdekaym throughput, ... ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow registratio ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY videosupport=yes; Turn on support for SIP video ;disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=gsm nat=no localnet = 192.168.0.0 localmask = 255.255.255.0 register = 8003440:[EMAIL PROTECTED]/8003440 [sipgate1] type=friend username=8003440 secret=passwd host=sipgate.de nat=no canreinvite=no fromuser=8003440 fromdomain=sipgate.net qualify=yes dtmfmode=rfc2833 im open for any hints! regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von nicolas Gesendet: Sonntag, 9. Mai 2004 12:43 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Re: asterisk with german SIPGATE ? Hi, whats is your problem ? For me it works, but had problems too. nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cron job to reboot GS101
On Sat, Apr 03, 2004 at 02:36:59PM +1000, dkwok wrote: Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? You might use curl for regular reboot as described in http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke!
Mark, Would you please re-config or use a different mail client as to not send your replies back as attachments?? It's VERY kludgy, and, I'm just going to stop reading them.. along with all the other folks.. Thanks, Billy - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:41 AM Subject: Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Re: asterisk with german SIPGATE ?
I´ve behind a firewall (Linux FLI4L) - but i have configured all possible Forwardings. Two Problems at this time: a) after many tries I have registered on siptel *CLI sip show registry Host Username Refresh State 217.10.79.9:5060 8003440 120 Registered I can make calls - but I can´t hear anything! -snip- I think the problem is that the RDP is not coming to the Asterisk? SIPGate-Website shows me as online! -snip- My Router is 0n 192.168.0.1, my Asterisk Server is 192.168.0.105 Here are my portforwardings: PORTFW_10='5004 192.168.0.105 UDP' # Port für SIP (RTP) zum Debian PORTFW_11='5060 192.168.0.105 tcp' # Port für SIP TCP zum Debian PORTFW_12='5060 192.168.0.105 udp' # Port für SIP UDP zum Debian PORTFW_13='5060 192.168.0.105 tcp' # Port für SIP UDP zum Debian PORTFW_14='5070-5080 192.168.0.105 udp'# für das RTP vom SIP-Phone PORTFW_24='8000-8012 192.168.0.105 udp' # SIPGATE ?!?! PORTFW_25='1-2 192.168.0.105 udp'# SIP ?!?! The problem is directly assoicated with rtp traffic for sure. The sip register function happens across port 5060 just fine because your system initiates that conversion, and your firewall allows the conversation because you initiated. The rtp traffic (voice) uses negotiated udp ports that are not very predicatable, are not standard ports between different devices, and can be changed by the person controlling the equipment at either end. About the only realistic way to get a handle on exactly which ports are attempted is to install a packet sniffer (like ethereal) and look at what ports are trying to be used. (This has been discussed many many times on the list, and you should be able to find hundreds if not thousands of references to it as well as on the wiki.) Asterisk attempts to use udp ports between 10,000 and 20,000 (defined in rtp.conf file), Cisco 79x0 phones between ports 16,384 and 32766, etc. Given that I see 8000-8012 in the above table, I assume your trying to use xten as well. Not sure if they still use those ports or not. The problem is basically related to the distant end attempting to contact your asterisk using some unknown/undocumented port, and your firewall or router is blocking that (as it should be). For testing purposes, open up all inbound udp ports from 5000 to 5, and then play around with the nat configuration statements in your sip.conf. The more appropriate way to do this really is to use ethereal to determine the exact ports needed as mentioned, and only open those up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AW: Re: asterisk with german SIPGATE ?
Hi, so i would do following: 1. you type your external ip into the sip.conf externip=x.x.x.x 2. nat=yes Your forwarding are ok, think you need udp only : 5060 and 1-2 see it in your rtp.conf. And you need open your ports to. should work. nico Thorsten Gehrig wrote: Hi experts (hope so), I?ve behind a firewall (Linux FLI4L) - but i have configured all possible Forwardings. Two Problems at this time: a) after many tries I have registered on siptel *CLI sip show registry Host Username Refresh State 217.10.79.9:5060 8003440 120 Registered I can make calls - but I can?t hear anything! Asterisk shows: -- Executing Dial(SIP/thorstengehrig-2641, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- SIP/sipgate1-dd8a answered SIP/thorstengehrig-2641 -- Attempting native bridge of SIP/thorstengehrig-2641 and SIP/sipgate1-dd8a I think the problem is that the RDP is not coming to the Asterisk? SIPGate-Website shows me as online! b) the second Problem is that Phonecalls to Sipgate are not forwarded to my Asterisk (I cant see anything on the console). (but i?m in Registered state and Website shows me online). I?ve configurated the sip-parts with qualify=yes and I recive this information: *CLI sip show peers Name/usernameHost Mask Port Status thorstengehrig/ 192.168.0.100 (D) 255.255.255.255 5060 OK (2 ms) sipgate1/800344 217.10.79.9 255.255.255.255 5060 OK (118 ms) My Router is 0n 192.168.0.1, my Asterisk Server is 192.168.0.105 Here are my portforwardings: PORTFW_10='5004 192.168.0.105 UDP' # Port für SIP (RTP) zum Debian PORTFW_11='5060 192.168.0.105 tcp' # Port für SIP TCP zum Debian PORTFW_12='5060 192.168.0.105 udp' # Port für SIP UDP zum Debian PORTFW_13='5060 192.168.0.105 tcp' # Port für SIP UDP zum Debian PORTFW_14='5070-5080 192.168.0.105 udp'# für das RTP vom SIP-Phone PORTFW_24='8000-8012 192.168.0.105 udp' # SIPGATE ?!?! PORTFW_25='1-2 192.168.0.105 udp'# SIP ?!?! here is my SIP.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls srvlookup = no ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel tos=lowdelay; Type of Service kiwdekaym throughput, ... ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow registratio ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY videosupport=yes; Turn on support for SIP video ;disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=gsm nat=no localnet = 192.168.0.0 localmask = 255.255.255.0 register = 8003440:[EMAIL PROTECTED]/8003440 [sipgate1] type=friend username=8003440 secret=passwd host=sipgate.de nat=no canreinvite=no fromuser=8003440 fromdomain=sipgate.net qualify=yes dtmfmode=rfc2833 im open for any hints! regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von nicolas Gesendet: Sonntag, 9. Mai 2004 12:43 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Re: asterisk with german SIPGATE ? Hi, whats is your problem ? For me it works, but had problems too. nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip to PSTN Gateway Configs
Bob, What I am going to tell you may seem arrogant or what, but I think you would do yourself a great favor if you figured this one out yourself by studying the info that is available and ask questions if things don't work. Your configuration is indeed very simple and with 100% certainty you will want more once you have it, and you will come back for more sample configurations, but haven't learned the basics to simply build on what you got. This response is really addressed to all that are starting out with Asterisk. With Asterisk is pays well (in terms of satisfaction and total time invested) to spend--no INVEST--some startup time to learn the basics, starting very simple and build up a system and dial plan step by step. I am seeing over and over here and other boards that starters want everything right now in their first configuration and therefore download all kinds of configuration files from the web and are confused by it all, because nothing fits together, and the simple principles are lost among the complexity. If one follows that advise, Asterisk is really very simple to set up. bob mc wrote: Hi, I'm trying to put together a simple gateway configuration involving Asterisk. I have a machine with 2 Digium X100P FXO cards installed and the Asterisk Software, and I have 2 Sip Phones defined. What I want to achieve is, any call arriving at FXO 1 is forwarded to Sip phone 1 only, and any call received on FXO 2 is transferred to Sip Phone 2, conversely any call originating from Sip Phone 1 goes out of FXO 1, and any call originating from Sip Phone 2 goes out of FXO 2. Some example .conf files would be greatly appreciated. Thanks Yahoo! Messenger - Communicate instantly...Ping your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stripping numbers at the end of a dial pattern = extension
Hello,
From: Hermann Wecke [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern =
extensions.conf
Date: 8 May 2004 22:03:57 +
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
Use ${variable:pos:n}. This will give you 'n' digits from the position
'pos'.
exten = 12345, 1, SetVar,MYDIGITS=${EXTEN:2,3} ; MYDIGITS = 2345.
Also, there is Substring application available with Asterisk, but it is
deprecated i think...
HTH, Girish
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[Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and inbound
calls are working fine (from both: Carrier and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel = 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel = 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
-- Making new call for cr 32774
Protocol Discriminator: Q.931 (8) len=40
Call Ref: len= 2 (reference 6/0x6) (Originator)
Message type: SETUP (5)
Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
Display (len= 6) [ 1Felix ]
Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '812' ]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
Sending Complete (len= 0)
-- Called 1/899312
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: STATUS (125)
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Info. element nonexist or not implemented
(99), class = Protocol Error (6) ]
Cause data 0: 14 (20)
Cause data 1: 01 (1)
Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state:
Call Initiated (1)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=13
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Invalid number format (28), class = Normal
Event (1) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Public network serving the local user (2)
Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (Cause)
-- Processing IE 30 (Progress Indicator)
-- Channel 1, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 6/0x6) (Originator)
Message type: RELEASE (77)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
RE: [Asterisk-Users] Stripping numbers at the end of a dial pattern = extension
Hi,
Replying to my own mail. There is a mistake, The syntax is incorrect:
From: Girish Gopinath [EMAIL PROTECTED]
exten = 12345, 1, SetVar,MYDIGITS=${EXTEN:2,3} ; MYDIGITS = 2345.
Correct: exten = 1234, 1, SetVar,MYDIGITS=${EXTEN:2:3} ; MYDIGITS = 234.
My apologies...
Girish
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Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke!
Billy, Attachment seems to be due to a GNUPG sig file -- William On Sun, 2004-05-09 at 12:00, Billy Huddleston wrote: Mark, Would you please re-config or use a different mail client as to not send your replies back as attachments?? It's VERY kludgy, and, I'm just going to stop reading them.. along with all the other folks.. Thanks, Billy - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:41 AM Subject: Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 seconds delay with Macros
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 07 May 2004 09:44 am, Uriel Carrasquilla wrote: I have noticed that when I switched to macros in my extensions.conf, there is now a 5 second delay. The macro starts with an announcement and then voicemail. Has anybody noticed the same? is it a feature? URiel ANOTHER REMINDER NOT TO USE REPLY FOR NEW MESSAGES Please note that this mailing list uses threading which allows us to track each issue per thread. When you press reply to send a message to the list it gets inserted into someone elses thread, like the one above. This may not be evident to those not fortunate enough to have threading in their mail client, but for the rest of us it's very annoying. This can be avoided quite simply by using the New Message To (or similarly marked) instead of Reply. This will start a new thread for Your new subject. Thanks, - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAnnUSljK16xgETzkRAuTrAJ0e7A5/cItOUjx1yc5+GxC9egXx4gCgqSCr qzsgUbm+FXiAT0F0J0NbXUk= =5YcZ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NOT USING REPLY TO THE LIST
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ANOTHER REMINDER NOT TO USE REPLY FOR NEW MESSAGES Please note that this mailing list uses threading which allows us to track each issue per thread. When you press reply to send a message to the list it gets inserted into someone elses thread, like the one above. This may not be evident to those not fortunate enough to have threading in their mail client, but for the rest of us it's very annoying. This can be avoided quite simply by using the New Message To (or similarly marked) instead of Reply. This will start a new thread for Your new subject. Thanks, - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAnnYlljK16xgETzkRAhVtAJ0YgKlY9ESpUq0W5D08L2aVA6dZbwCcCqhs RpAajsprAdrnfSRmL3ysY/Q= =TfPB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF broken
Some CVS upgrade in the last day or two has broken the recognition of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI -- Executing VoiceMailMain(SIP/phone1-e0dd, ) in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from [EMAIL PROTECTED] By re-installing an older (cvs checkout -r v1-0_stable asterisk) version - everything works fine again... thats with NO config changes at all.. Has someone removed some support for the transporting of DTMF (eg, info?) - I am using... dtmfmode=info in sip.conf with BudgeTone-100's (sent with absolutely no signatures or attachments) signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] DTMF broken
What firmware you have on that BT101? And yes gnupg or what ever you use to sign your message did produce the attachemnt on this last one too. bkw - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 12:23 PM Subject: [Asterisk-Users] DTMF broken ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF broken
Mark, Could you please add a SIP debug message with the SIP INFO? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: cron job to reboot GS101
I have used the cron job before, and it worked fine, but didn't seem to be more than a hack to me. I found that if I turned off the "Subscribe for MWI" flag in the GS config page, and it stopped losing registration. The MWI still works, it just stops sending the subscribe packets to the * box. YMMV. Brian --- Stefan Tichy <[EMAIL PROTECTED]> wrote: On Sat, Apr 03, 2004 at 02:36:59PM +1000, dkwok wrote: Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? You might use curl for regular reboot as described in http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager - inconsistent use of [--END COMMAND--]
All: Why do some requests to the manager return [--END COMMAND--] and some don't? (version 0.7.1) In the following example show version has it and sip show peers doesn't. Why? Thanks for any suggestions! [Action: Command, Command: sip show peers] [Response: Follows] [Name/usernameHost Mask Port Status] [2004/2004192.168.0.8 (D) 255.255.255.255 5060 Unmonitored] [2003/2003192.168.0.6 (D) 255.255.255.255 5060 Unmonitored] [2002/2002(Unspecified) (D) 255.255.255.255 0 Unmonitored] [2001/2001(Unspecified) (D) 255.255.255.255 0 Unmonitored] [2000/2000(Unspecified) (D) 255.255.255.255 0 Unmonitored] [iconnect/253735 213.137.73.140 255.255.255.255 5060 Unmonitored] [Action: Command, Command: show version] [Response: Follows] [Asterisk 0.7.2 built by [EMAIL PROTECTED] on a i686 running Linux] [--END COMMAND--] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ePyron Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without
alarms and inbound calls are working fine (from both: Carrier
and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel = 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel = 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
-- Making new call for cr 32774
Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2
(reference
6/0x6) (Originator) Message type: SETUP (5) Bearer
Capability (len= 3)
[ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode
(16)
Ext: 1 User information layer
1: A-Law
(35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ] Display (len= 6)
[ 1Felix ]
Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '812' ]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
Sending Complete (len= 0)
-- Called 1/899312
Protocol Discriminator: Q.931 (8) len=14 Call Ref: len=
2 (reference 32774/0x8006) (Terminator) Message type: STATUS (125)
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Info. element nonexist or
not implemented
(99), class = Protocol Error (6) ]
Cause data 0: 14 (20)
Cause data 1: 01 (1)
Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard
(0) Call state:
Call Initiated (1)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
Protocol Discriminator: Q.931 (8) len=10 Call Ref: len=
2 (reference 32774/0x8006) (Terminator) Message type: CALL
PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID:
Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (Channel Identification) Protocol
Discriminator: Q.931 (8) len=13 Call Ref: len= 2
(reference
[Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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[Asterisk-Users] Cannot Dial out with xp100
PGPexch.htm.pgp Description: Binary data
Re: [Asterisk-Users] DTMF broken
On Sun, 2004-05-09 at 20:48, Olle E. Johansson wrote: Mark, Could you please add a SIP debug message with the SIP INFO? I've done a debug with a working asterisk (V1.0) and the non-working asterisk. The trace is attached. :-)(debug - ascii text) When you say SIP INFO - what else are you asking for??? If its one of the 'sip show' commands - which one, and at what instance of time? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 This is a debug trace of Asterisk v1-0_stable - I'm dialing '310' which in extensions.conf looks like.. ; 310 = Access Voicemail - with full prompting exten = 310,1,VoicemailMain() I'm hanging up after 'dialing' 203 ... the 'bad' one follows after *CLI sip debug SIP Debugging Enabled *CLI Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2408 INVITE User-Agent: Grandstream BT100 1.0.4.63 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 345 v=0 o=phone1 8000 8000 IN IP4 160.124.48.121 s=SIP Call c=IN IP4 160.124.48.121 t=0 0 m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:40 12 headers, 16 lines Using latest request as basis request Sending to 160.124.48.121 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format GSM Found audio format UNKN Found description format iLBC Found description format PCMU Found description format PCMA Found description format G729 Found description format G722 Found description format G723 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 1309/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone;tag=as3564c06e Call-ID: [EMAIL PROTECTED] CSeq: 2408 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=6d4d7372 Content-Length: 0 to 160.124.48.121:5060 Sip read: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone;tag=as3564c06e Contact: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2408 ACK User-Agent: Grandstream BT100 1.0.4.63 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6 From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED];user=phone Proxy-Authorization: DIGEST username=phone1, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED] a;user=phone, nonce=6d4d7372, response=0142fb85eda2d7497992a0149d78e828 Call-ID: [EMAIL PROTECTED] CSeq: 2409 INVITE User-Agent: Grandstream BT100 1.0.4.63 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 345 v=0 o=phone1 8000 8000 IN IP4 160.124.48.121 s=SIP Call c=IN IP4 160.124.48.121 t=0 0 m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:40 13 headers, 16 lines Using latest request as basis request Sending to 160.124.48.121 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format GSM Found audio format UNKN Found description format iLBC Found description format PCMU Found description format PCMA Found description format G729 Found description format G722 Found description format G723 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 1309/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0
RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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Re: [Asterisk-Users] DTMF broken
On Sun, 2004-05-09 at 21:39, brian k. west wrote: What firmware you have on that BT101? And yes gnupg or what ever you use to sign your message did produce the attachemnt on this last one too. OK the gnuPG is off.. :-( Product Model:BT100 Software Version: Program--1.0.4.63 Bootloader--1.0.0.16 HTML--1.0.0.30 VOC--1.0.0.5 -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Assistance
That is not working...
I tried like you mentioned it and even a few different ways and will not
create the file at all
Am I doing something wrong...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09, 2004 4:00 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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Re: [Asterisk-Users] Where to start?
Ed Mansouri wrote: Hello, The goal would be to minimize the expenditure on new equipment, and utilize as much of the existing equipment as possible, so I guess it just depends on what the requirements of working with an open source PBX are. The current desktop phones are pretty sophisticated and provide a lot of functionality the preference would be to continue to employ them. It looks like that at a minimum, you'll need at least one 4-port FXO card (which Digium now sells...for around $350 I think). This would handle the incoming calls. Most proprierary PBX systems like to use their own phones as well, so in almost all cases they are not going to work w/ anything outside the PBX they were designed for. That's just how it is...and how PBX vendors can lock you into their technology. You could set the PBX up to pass-through calls on the analog lines with a 4-port FXS card ($305 @ digium)...and extend w/ IP phones as you can move things over. As far as phones go, you can find Cisco 7960s online for as little at $400 apice...or you can deploy softphones to your desktop computer and use those instead. -- /* Ian A. Underwood - [EMAIL PROTECTED] - http://www.agentgreen.org There are 4 boxes to use in the defense of liberty: soap, ballot, jury, ammo. Use in that order. Starting now. */ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy problem?!?
I have the uhci_usb modules etc. installed. Make sure this is a module and *not* part of the Kernel. I have usb-uhci and usbcore as modules. What about PPP support? Is that a problem? Should I also install it as a module? regards, thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] German sound files available
Hi there, today I made the German language prompts available for download: http://www.karl.aegee.org/asterisk.nsf/HT/sound-de Be aware: Asterisk doesn't yet fully support languages other than English, there are still (smaller) issues with voicemail and date/time announcements that require a patch. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Assistance
Never Mind... Figured it out... Thanks...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09, 2004 4:00 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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RE: [Asterisk-Users] AGI Assistance
Can you post your error to the list so we know what was wrong?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Sent: Monday, 10 May 2004 7:34 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] AGI Assistance
Never Mind... Figured it out... Thanks...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09, 2004 4:00 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
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[Asterisk-Users] Help with initial setup
Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Me 2124 [phone2] type=friend ;secret=blah host=dynamic defaultip=192.168.1.107 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Mini Me 2123 And in extensions.conf at the very end: [sip] exten = 1,1,Dial(SIP/phone1,20,tr) exten = 2,1,Dial(SIP/phone2,20,tr) exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr) These are budgetone 102's, so I've then proceeded to their admin interface, and told them that the sip server is: 192.168.1.13. For phone1, all I've set is the sip id/username as phone1 and likewise for phone2 on phone number two. Rebooted.. But I do not seem to be able to get them to talk to asterisk. When issuing a sip show peers in asterisk, it displays: Name/usernameHost Mask Port Status phone2/phone2192.168.1.107 (D) 255.255.255.255 5060 Unmonitored phone1 (Unspecified) (D) 255.255.255.255 0Unmonitored And when a sip show registry is issued, nothing seems to be connected: Host Username Refresh State Could there be something I'm missing in order to get the very basic working and then expand on that? Thanks in advance. Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk webmin
Title: Nachricht Hello, i have found a webmin module on the astersik ftp server! but how can i install it?! Can anybody help me! thanks in advance best regards markus Dohnal
Re: [Asterisk-Users] Help with initial setup
Phone two is registering ok, double check phone one against phone two's settings. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 3:51 PM Subject: [Asterisk-Users] Help with initial setup Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Me 2124 [phone2] type=friend ;secret=blah host=dynamic defaultip=192.168.1.107 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Mini Me 2123 And in extensions.conf at the very end: [sip] exten = 1,1,Dial(SIP/phone1,20,tr) exten = 2,1,Dial(SIP/phone2,20,tr) exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr) These are budgetone 102's, so I've then proceeded to their admin interface, and told them that the sip server is: 192.168.1.13. For phone1, all I've set is the sip id/username as phone1 and likewise for phone2 on phone number two. Rebooted.. But I do not seem to be able to get them to talk to asterisk. When issuing a sip show peers in asterisk, it displays: Name/usernameHost Mask Port Status phone2/phone2192.168.1.107 (D) 255.255.255.255 5060 Unmonitored phone1 (Unspecified) (D) 255.255.255.255 0 Unmonitored And when a sip show registry is issued, nothing seems to be connected: Host Username Refresh State Could there be something I'm missing in order to get the very basic working and then expand on that? Thanks in advance. Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hi Felix,
on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:
pridialplan=national
Have you tried:
pridialplan=unknown
in zapata.conf?
It seems as though the omission of the pridialplan= statement in zapata.conf
is treated by Asterisk as pridialplan=national.
We could probably give you more relevant suggestions if you would enable a
more verbose level of output and post the call setup trace results here. Try
the following command from the Asterisk CLI before making your next call:
pri debug span x
Where x = single integer digit for the PRI span that will be used to make
the outgoing call. (Eg. 1)
Please drop a note to the list (either way) with your results.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ePyron Felix
Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ePyron Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without
alarms and inbound calls are working fine (from both: Carrier
and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel = 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel = 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
-- Making new call for cr 32774
Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2
(reference
6/0x6) (Originator) Message type: SETUP (5) Bearer
Capability (len= 3)
[ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode
(16)
Ext: 1 User information layer
1: A-Law
(35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ] Display (len= 6)
[ 1Felix ]
Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '812' ]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
Sending Complete (len= 0)
-- Called 1/899312
Protocol Discriminator: Q.931 (8) len=14 Call Ref: len=
2 (reference 32774/0x8006) (Terminator) Message type: STATUS
Re: [Asterisk-Users] Low Bit Rate Codecs
Thanks Steve, that was good reading. - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 08, 2004 9:21 PM Subject: Re: [Asterisk-Users] Low Bit Rate Codecs Craig wrote: Greetings all, I have searched all over and have found bits and pieces on low bit rate codecs, however I have found it very difficult to compare apples with apples. The conclusions I have come to are as follows, I would appreciate if anyone has some feedback, or point me to where I might find this sort of comparison in black and white G723.1 very low bit rate used commercially, not avail for * (I am currently using this codec in another commercial application and therefore it is my reference point) G.723.1 is pretty much obsolete. You don't see it being used on anything new. Most people do VoIP using RTP. The overhead of RTP is so huge, a small saving on the codec makes little difference. People generally go for G,729 now, which sounds considerably better. If you compare the total bit rate for G.723.1 vs G.729 in RTP G.723.1 often comes out considerably lower. This is because it works in 30ms blocks - you only have 33 RTP packet overheads per second. You can choose to pack more G.729 data into each RTP packet and even this up. The patent licencing for G.723.1 is a PITA, which hasn't helped it achieve widespread use. There are two variants of G.723.1, with different bit rates. The lower bite rate (5.something kbps) sounds nasty. The higher rate (6.something kbps) sounds more reasonable. Using 30ms blocks, it is not so compatible with *, which is geared to 20ms block processing. A lost packet causes a 30ms hole, so it tends to be less tolerant of packet loss than something working in smaller blocks. It sounds awful for anything but a single pure voice. G729a low bit rate slightly higher bandwidth usage than 723.1 ??? avail as a low cost add-on for * better quality that g723.1 ??? Definitely better quality than G.723.1. This is definitely the mainstream right now for VoIP. It is heavily patented, so free codecs are not possible. There are several bit rate options, but almost everyone uses the 8kbps variant. This sounds pretty good for its bit rate, though I think there have been better codecs. In telephony you need use something compatible with the far end, and G.729 seems to be the current common ground. It is rather intolerant of packet loss. Some people pack several G.729 blocks into a single RTP packet, to decrease the RTP overhead. That makes it even less tolerant of packet loss. It sounds awful for anything but a single pure voice. iLBC Low bit rate slightly higher bandwidth usage than 723.1 and 729a ??? open source, no additional cost for * quality comparable to G729a stands up better ip paths suffering from latency and jitter ??? iLBC has a much higher bit rate than G.729, but the voice quality is about the same. Why does that make it interesting? Well, it is designed to be much more tolerant of packet loss, and that makes it take more bandwidth. The design of RTP makes that take so much overhead that the total bit rate using iLBC isn't a huge jump from using G.729. However, if you use a more efficient streaming mechanism - say IAX, or an RTP like format with many calls packed in a packet - the total bit rate difference starts to look wider. There, the increase in bits is so great its quite likely to be the *cause* of packet loss, by clogging up the channel. :-) Good old GSM 06.10 is worthy of consideration. Free of patents (at least ones being actively pursued). Low compute requirements. Reasonable voice quality. Somewhat more tolerant of background noise than the codecs above. Although GSM networks don't use it much these days (they mostly use the newer EFR and half rate codecs) it's still a very servicable codec. Its bit rate lies between G.729 and iLBC. On a pure voice it gives poorer quality than G.729. Add some background noise and it can beat G.729. Its tolerance of packet loss is probably similar to G.729. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-oh323, compile problems using V0.6.0 or 0.6.1
Thanks for that, the patch got me a little further.
I am now getting this error during the asterisk-oh323 compile.
Any ideas.
mipt:/usr/src/asterisk-oh323-0.6.1 # make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper'
./check_ver /usr/src/pwlib pwlib
./check_ver /usr/src/openh323 openh323
g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC
-I/usr/src/pwlib/include -DPTRACING -I/usr/src/openh323/include -DHAS_OSS -Wall
-DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.7.0\
-DOPENH323VERSION=\1.14.0\ -I/usr/include/openssl
-I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include
-I/usr/src/openh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver
-x c++ -Os -g -c wrapendpoint.cxx -o wrapendpoint.o
wrapendpoint.cxx: In member function `virtual BOOL
WrapH323EndPoint::OpenAudioChannel(H323Connection, int, unsigned int,
H323AudioCodec)':
wrapendpoint.cxx:717: error: `IsDescendant' undeclared (first use this
function)
wrapendpoint.cxx:717: error: (Each undeclared identifier is reported only once
for each function it appears in.)
/usr/include/g++/iostream: At top level:
/usr/src/openh323/include/h323pluginmgr.h:55: warning: `
H323PluginCodec_PluginLoader*H323PluginCodec_PluginLoader_Static' defined
but not used
make[1]: *** [wrapendpoint.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.1/wrapper'
make: *** [subdirs_all] Error 1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David
Hindmarsh
Sent: Friday, 7 May 2004 3:41 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisk-oh323, compile problems using V0.6.0
or 0.6.1
Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as
recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I get the following errors, I have checked that the
envorinment variables are set correctlty as below.
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib
g++ (GCC) 3.3.1 (SuSE Linux)
The errors from the compile are below
mipt:/usr/src/asterisk-oh323-0.6.1 # make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target `ccflags'. Stop.
make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper'
./check_ver /usr/src/pwlib pwlib
./check_ver /usr/src/openh323 openh323
g++ -Wall -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.7.0\
-DOPENH323VERSION=\1.14.0\ -I/usr/include/openssl
-I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include
-I/usr/src/openh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver
-x c++ -Os -g -c wrapper_misc.cxx -o wrapper_misc.o
In file included from /usr/src/pwlib/include/ptlib.h:172,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error: parse error before
`protected'
/usr/src/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error: syntax error
before `*' token
In file included from /usr/src/pwlib/include/ptlib.h:184,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:53: error: parse error before
`public'
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:55: error: destructors must be
member functions
/usr/src/pwlib/include/ptlib/unix/ptlib/config.h:57: error: parse error before
`protected'
In file included from /usr/src/pwlib/include/ptlib.h:190,
from wrapper_misc.hxx:35,
from wrapper_misc.cxx:34:
/usr/src/pwlib/include/ptlib/args.h:121: error: parse error before `{' token
/usr/src/pwlib/include/ptlib/args.h:147: error: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:156: error: parse error before `const'
/usr/src/pwlib/include/ptlib/args.h:165: error: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:175: error: parse error before `int'
/usr/src/pwlib/include/ptlib/args.h:190: error: `ostream' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:191: error: `strm' was not declared in this
scope
/usr/src/pwlib/include/ptlib/args.h:191: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:191: error: variable or field `PrintOn'
declared void
/usr/src/pwlib/include/ptlib/args.h:197: error: `istream' was not declared in
this scope
/usr/src/pwlib/include/ptlib/args.h:198: error: `strm' was not declared in this
scope
/usr/src/pwlib/include/ptlib/args.h:198: error: virtual outside class
declaration
/usr/src/pwlib/include/ptlib/args.h:198: error: variable or field `ReadFrom'
declared void
/usr/src/pwlib/include/ptlib/args.h:206: error: parse error before `' token
Re: [Asterisk-Users] Asterisk webmin
On Mon, 10 May 2004, Administrator wrote: i have found a webmin module on the astersik ftp server! It is broken. Forget it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Bit Rate Codecs
nasty. The higher rate (6.something kbps) sounds more reasonable. Using 30ms blocks, it is not so compatible with *, which is geared to 20ms block processing. A lost packet causes a 30ms hole, so it tends to be less tolerant of packet loss than something working in smaller blocks. It sounds awful for anything but a single pure voice. * can do 30ms codecs. iLBC is one such codec... bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ztdummy problem?!?
What kernel version?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Schroeter Sent: Sunday, May 09, 2004 4:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ztdummy problem?!? I have the uhci_usb modules etc. installed. Make sure this is a module and *not* part of the Kernel. I have usb-uhci and usbcore as modules. What about PPP support? Is that a problem? Should I also install it as a module? regards, thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello Again Felix,
first a quick apology: sorry, I re-read your e-mail and found the trace
information (lower down) that you had already posted. (It's late here, etc.)
The error messages that you reported in your last e-mail are actually
outbound Q.931 call setup messages that are being sent to DTAG from your
Asterisk machine. The direction of the message is indicated in the first
column of the trace output in the form of or . Although these are not
error messages I am surprised to see those particular messages being
generated with your current zapata.conf settings; with pridialplan=local I
would have expected something similar to the following messages during call
setup:
Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) 'X58777' ]
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'X986600' ]
(I have inserted X in the PSTN numbers above to protect the innocent
Calling and Called parties.)
Please retry pridialplan=local and pridialplan=unknown in zapata.conf and
post the trace results so we compare results. With pridialplan=local in
zapata.conf the outbound call setup from Asterisk to DTAG should look ideal.
On a different subject, how are your results with telephony calls from the
Asterisk machine to your Hicom PBX? I would have expected the zaptel.conf
entry to have been:
#hicom (siemens)
span=2,0,0,ccs,hdb3,crc4
...so that your Asterisk provides clocking/timing information for the Hicom.
If this configuration is not set correctly you could find that the systems
seem to communicate well at first but after a while you might see strange
PRI errors (every hour or so) that relate to clock synchronisation problems.
MfG
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storer,
Darren
Sent: 10 May 2004 01:29
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hi Felix,
on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:
pridialplan=national
Have you tried:
pridialplan=unknown
in zapata.conf?
It seems as though the omission of the pridialplan= statement in zapata.conf
is treated by Asterisk as pridialplan=national.
We could probably give you more relevant suggestions if you would enable a
more verbose level of output and post the call setup trace results here. Try
the following command from the Asterisk CLI before making your next call:
pri debug span x
Where x = single integer digit for the PRI span that will be used to make
the outgoing call. (Eg. 1)
Please drop a note to the list (either way) with your results.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ePyron Felix
Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ePyron Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without
alarms and inbound calls are working fine (from both: Carrier
and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[Asterisk-Users] Help!! Music On Hold
I've been trying to play the default music on hold file, but no luck yet. here is my configuration: extensions.conf [incoming] exten = s,1,Dial,Zap/2|10 exten = s,2,Voicemail,u34 exten = s,102,Voicemail,b34 exten = 34,1,SetMusicOnHold,default Musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered = mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3 zapata.conf musiconhold=default Any solution to this issue? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help!! Music On Hold
It's obvious you've at least tried to figure it out since you've used the SetMusicOnHold app, so I'll be nice. Try MusicOnHold() http://www.voip-info.org/wiki-Asterisk+cmd+MusicOnHold -Original Message- From: leonardo [mailto:[EMAIL PROTECTED] Sent: Sunday, May 09, 2004 9:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help!! Music On Hold I've been trying to play the default music on hold file, but no luck yet. here is my configuration: extensions.conf [incoming] exten = s,1,Dial,Zap/2|10 exten = s,2,Voicemail,u34 exten = s,102,Voicemail,b34 exten = 34,1,SetMusicOnHold,default Musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered = mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3 zapata.conf musiconhold=default Any solution to this issue? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk webmin
well, you have to have webmin installed first. you can get webmin from www.webmin.com. Once you have that done, you can install the module into webmin by logging into webmin with a browser, going into the webmin management section then to the modules section and it become pretty obvious there. Administrator wrote: Hello, i have found a webmin module on the astersik ftp server! but how can i install it?! Can anybody help me! thanks in advance best regards markus Dohnal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Assistance
The error was in the Quotes and Date Variable
The end result looks as follows...
$Date = time();
$Path = /usr/local/apache/htdocs/demo/sound/$EmpNum.$Date;
$ShortPath = sound/$EmpNum.$Date;
$AGI-exec('Record', $Path:wav);
I needed the $ShortPath variable for some web values... But besides
that This is what did it for me
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Posted At: Sunday, May 09, 2004 6:40 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Can you post your error to the list so we know what was wrong?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Sent: Monday, 10 May 2004 7:34 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] AGI Assistance
Never Mind... Figured it out... Thanks...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman Posted At: Sunday, May 09, 2004 4:00 PM Posted To: Asterisk
User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE: [Asterisk-Users] AGI Assistance
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI Assitance
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
so I can store it in a database.
Any thoughts
Thanks,
-gcc
___
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Asunto: Re: [Asterisk-Users] Asterisk webmin
Where i can download the module ?? regards Ivan -- Mensaje original -- From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk webmin Reply-To: [EMAIL PROTECTED] Date: Sun, 09 May 2004 19:30:33 -0700 well, you have to have webmin installed first. you can get webmin from www.webmin.com. Once you have that done, you can install the module into webmin by logging into webmin with a browser, going into the webmin management section then to the modules section and it become pretty obvious there. Administrator wrote: Hello, i have found a webmin module on the astersik ftp server! but how can i install it?! Can anybody help me! thanks in advance best regards markus Dohnal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asunto: Re: [Asterisk-Users] Asterisk webmin
um its in ftp.asterisk.org but its BROKEN bkw - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:53 PM Subject: Asunto: Re: [Asterisk-Users] Asterisk webmin Where i can download the module ?? regards Ivan -- Mensaje original -- From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk webmin Reply-To: [EMAIL PROTECTED] Date: Sun, 09 May 2004 19:30:33 -0700 well, you have to have webmin installed first. you can get webmin from www.webmin.com. Once you have that done, you can install the module into webmin by logging into webmin with a browser, going into the webmin management section then to the modules section and it become pretty obvious there. Administrator wrote: Hello, i have found a webmin module on the astersik ftp server! but how can i install it?! Can anybody help me! thanks in advance best regards markus Dohnal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail: upgraded?
Look at bugs.digium.com - Search for Voicemail... You will find it there... Geoff Clark Network Engineer The Network Essentials [EMAIL PROTECTED] 704-568-0031 (W) 704-622-3905 (C) www.tnessentials.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Friday, May 07, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail: upgraded? I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asunto: Re: [Asterisk-Users] Asterisk webmin
Just downloaded the file caleld webmin.tgz. It appears to be a snapshot of someones experimental stuff. It's mis-named to be able to install per my instructions (my apologies all) and as structured webmin's module installer won't. I'll see if I can get it into a form where it will install into webmin. If I do get it installable, who should I send it to for testing against an asterisk system as I haven't one here. brian k. west wrote: um its in ftp.asterisk.org but its BROKEN bkw - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:53 PM Subject: Asunto: Re: [Asterisk-Users] Asterisk webmin Where i can download the module ?? regards Ivan -- Mensaje original -- From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk webmin Reply-To: [EMAIL PROTECTED] Date: Sun, 09 May 2004 19:30:33 -0700 well, you have to have webmin installed first. you can get webmin from www.webmin.com. Once you have that done, you can install the module into webmin by logging into webmin with a browser, going into the webmin management section then to the modules section and it become pretty obvious there. Administrator wrote: Hello, i have found a webmin module on the astersik ftp server! but how can i install it?! Can anybody help me! thanks in advance best regards markus Dohnal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems when upgraded
I have just installed one of the new TDM400 cards with an FXS and an FXO module into my * server. I also checked out the latest cvs head. I am using 7940 phones. Now I have some strange problems: 1. When in the VM menus, key presses do not register. 2. When I press hold on the 7940, it hangs up. Has anyone got any ideas? TIA Simon Brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Example: calling card using extension logic ONLY!
http://www.bkw.org/~brian/callingcard.conf Now this is far from a complete example but it doesn't allow more than 1 person to use the card at the same time(thats right only one person at a time can use the card). Its just an example and shouldn't be used in production. This is only to show how you can do complicated operations using only extension logic. The sounds are from asterisk-sounds. They are far from the best ones for this but hey I just wanted to prove it could be done and show a use of app_groupcount NEXT!!! bkw
Re: [Asterisk-Users] Voicemail: upgraded?
Its in CVS HEAD ... check /usr/src/asterisk/configs/voicemail.conf.sample bkw - Original Message - From: AstGrp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 9:08 PM Subject: RE: [Asterisk-Users] Voicemail: upgraded? Look at bugs.digium.com - Search for Voicemail... You will find it there... Geoff Clark Network Engineer The Network Essentials [EMAIL PROTECTED] 704-568-0031 (W) 704-622-3905 (C) www.tnessentials.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Friday, May 07, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail: upgraded? I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905 vs Cisco 7905-G
Mathieu Nantel wrote: Hey, Can anyone comment on the difference between the 7905 and it's upgrade, the 7905-G ? Has anyone used these phones in a configuration? Is SIP well implemented in the 7905 ? Thanks in advance, Mat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users in sip and h323 working goot options G is power ethernet version must send in cable 48V ~ I think see on cisco web side ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stripping numbers at the end of a dial pattern = extensions.conf
http://www.voip-info.org/wiki-Asterisk+cmd+StripLSD
gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Posted At: Saturday, May 08, 2004 6:04 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Stripping numbers at the end of a dial
pattern = extensions.conf
Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern
= extensions.conf
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
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[Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
* Read the config sample files! (even if you're an Asterisk guru) - For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. As we add or change features in Asterisk, the sample config files are updated. If you look there, you might get new insights into how to solve your problems. Also, you might find new features that you really need. If you have a new installation gmake samples or make samples will install these files for you. In CVS head, the development source tree, we've added quite a lot of information recently to these files. They are more educational now and contains a lot of sample configurations. *** Check app_groupcount! - There's a new app in Asterisk town. In fact, there are several new applications in CVS head. One of the major recent additions is app_groupcount, that you can use to limit the number of calls to, well, just about anything. A SIP peer, a PRI link, a call center staff member, a conference and calls to or from your boy and/or girlfriend :-) The command for setting a group is setgroup(), the command for enforcing a limitation is checkgroup(). Please start using this instead of the incominglimit and outgoinglimit settings in sip.conf. These are not working as expected and the more general solution with app_groupcount is a much better solution that works cross channels. This is an end-of-life warning for outgoinglimit and incominglimit :-) As always, the CLI command show applications and show application name is your best friend. *** Set your SIP realm! --- In CVS head, the SIP channel is now able to use a proper SIP realm for authentication. The realm is the server group that has a common authentication for a user. It could be one server or a number of servers that shares a password/user database. According to the SIP RFC, it should be set either to a domain or a hostname, depending on what your realm covers. It should be globally unique. Up to know, all Asterisk servers used the asterisk realm. That made it a bit hard for some phones to know the difference between one server and another. Please note that if you are using the md5secret setting in sip.conf, this secret is based on the realm. If you change the realm, you need to rehash your secrets. *** Asterisk 1.0: Less than five bugs away -- If you follow the CVS, you will notice that there are very few changes in the stable part of the source tree. Only bug fixes go in there and Mark have been working like crazy to fix the major bugs. The bug tracker had almost 300 open bugs just a while ago, and we are now down to a handful identified bugs. As usual with Open Source Software, relase is not set to marketing plans. Release will come when the software is ready to be shipped. So when Mark decides that we've fixed the bugs that needs fixing, a release candidate will be made and published for download. Please plan to help us test the 1.0rc1 real hard. Do whatever you can to crash it, to make it dial your mother-in-law when you really want to talk to your husband, to make it connect the whole office to the HR departments secret conference call by mistake and accidentally fill your hard disk drive with non-existing voice mail messages. We do not belive that you can, but if you can, report the bugs and help us move forward to a 1.0 release! If you want to start stress-testing it now, download the stable CVS release. Instructions is to be found at http://www.asterisk.org *** Astricon: Coming right up, sir! --- We get a lot of questions about Astricon. To answer a few: - We're still open of speaker's proposals, even though the time limit is up. - The conference venue is not set yet, we will add it to the web site as soon as we have more information - Pre-registration will start rsn (real soon now) - We will find a location with a standard class hotel as well as a lower price alternative. - Yes, we will have the voice of Asterisk there (hint, hint) Astricon is at http://www.astricon.net * Useful pointers: -- * Asterisk: http://www.asterisk.org * Asterisk mailing lists: http://lists.digium.com (users, dev, biz and cvs mailing list) * Asterisk bug tracker: http://bugs.digium.com * Asterisk IRC channel: #asterisk on irc.freenode.net * Digium: http://www.digium.com * Wiki: http://www.voip-info.org * Voip Search: http://search.voip-forum.com * Astricon: http://www.astricon.net Have a nice Asterisk week! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
