[asterisk-users] Dial out trhough a FXS channel on a TDM card
Hello. I have a TDM 2400P and havent figured out how to attach a phone to one of the FXS channels in the bank and dial out. To dial in the analog phone is easy, all I had to do was to insert a line in the extensions.conf saying exten = 430,1,Dial(Zap/17,20,t). But I cant figure out how to have a line signal on the same phone to dial out. Anybody can help me please? Thanks, Robson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400P and Polycom phones
Dear All, thanks for the help on the TDM2400P. I have resolved the issue. I isolated the problem and ended up finding out it was the Polycom phone that had a problem. Those phones have spectacular quality but they are way too complicated to setup. Also, it's absurd Polycom only supplies you with the latest software if you ask through your reseller!!! What kind of rule is that? Is someone making Polycom phones in China other than Polycom. Well, in any case the phones got confused when selecting the right CODEC to use so I isolated Alaw andDONE. The TDM2400 is fine and works perfectly and so does the phone. One note to people trying to install these phones is that there is a guide on VOIP-Info for the IP500 (more expensive) but not for the Ip301, but they are the same. The other things is that this phone doesn't like to talk to other phones on different networks (like connecting your office and your home with the same phones. Robson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Friday, September 22, 2006 12:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM2400P On Thu, Sep 21, 2006 at 02:28:40PM -0300, Robson Ribeiro wrote: Dear Jay, maybe I would better describe the sound as breaking and not skipping. It is a constant thing so the person on the other side can't understand a word. It's like when you are in a bad cellphone connection. It ONLY happens and this is the weird part, when I call OUT of the TDM. When someone call IN nothing happens. The call is originating as a ZAP call on a FXSs channel and going directly to the PSTN. Now, I tried working with TX/RX But it didn???t make any difference as the issue doesn???t seem to matter if gain is higher or lower. If I was calling from a VOIP provider I could understand this as being a bandwidth issue. But from the PSTN to another PSTN it is very strange indeed. I tried calling you but noone answered. Will try later. I apologize; I failed to realize you were non-CONUS; the CNID was odd-looking, and I ignored the call. Feel free to try again. Can you try originating a call out your FXO port from a SIP phone? Is the audio ok when you call FXS to FXS? You need, in general, to use the process of elimination to figure out where your problem *can* be -- even if that entails borrowing hardware. Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400P
Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycoms IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = us defaultzone=us fxsks=1-4 fxsks=5-8 fxoks=17-20 fxoks=21-24 Zapata.conf [channels] language=en context=default ;switchtype=national echocancel=64 echocancelwhenbridged=no echotraining=800 toneduration=200 busydetect=yes signalling = fxs_ks rxgain=5.0 txgain=-10.0 channel = 1-4 channel = 5-8 signalling = fxo_ks channel = 17-20 channel = 21-24 Best Regards, Robson Ribeiro MSN: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400P
Dear Jay, maybe I would better describe the sound as breaking and not skipping. It is a constant thing so the person on the other side can't understand a word. It's like when you are in a bad cellphone connection. It ONLY happens and this is the weird part, when I call OUT of the TDM. When someone call IN nothing happens. The call is originating as a ZAP call on a FXSs channel and going directly to the PSTN. Now, I tried working with TX/RX But it didn’t make any difference as the issue doesn’t seem to matter if gain is higher or lower. If I was calling from a VOIP provider I could understand this as being a bandwidth issue. But from the PSTN to another PSTN it is very strange indeed. I tried calling you but noone answered. Will try later. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Thursday, September 21, 2006 1:23 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM2400P On Thu, Sep 21, 2006 at 12:28:16PM -0400, Robson Ribeiro wrote: Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycom’s IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. Well, if that weren't an analog card, I'd say it sounded like clock slip. It could be digital clipping/overdrive; you might check your gains. You have FXS, FXO, and SIP channels, there; which combinations cause the clicking in the transmit audio? Does it happen from FXS to FXO? SIP to FXO? SIP to SIP? How frequently, and how regularly, are the ticks? How loud? How sharp? Can you call someone with audio experience to describe them to you? (If no one else, feel free to call me; I'm good at this stuff... ;-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400P
Indeed there is something very strange here: look how the PC is recognizing the Digium boardIs this normal?also i have noticed that both the IVR and Musiconhold seem to be acceleratedafter lspci i get: :04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11)On 9/21/06, Robson Ribeiro [EMAIL PROTECTED] wrote:Dear Jay, maybe I would better describe the sound as breaking and not skipping. It is a constant thing so the person on the other side can't understand a word. It's like when you are in a bad cellphone connection. It ONLY happens and this is the weird part, when I call OUT of the TDM. When someone call IN nothing happens. The call is originating as a ZAP call on a FXSs channel and going directly to the PSTN. Now, I tried working with TX/RX But it didn't make any difference as the issue doesn't seem to matter if gain is higher or lower. If I was calling from a VOIP provider I could understand this as being a bandwidth issue. But from the PSTN to another PSTN it is very strange indeed. I tried calling you but noone answered. Will try later. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Jay R. AshworthSent: Thursday, September 21, 2006 1:23 PMTo: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] TDM2400P On Thu, Sep 21, 2006 at 12:28:16PM -0400, Robson Ribeiro wrote:Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs.They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, Ican hear perfectly but the person on the other end is hearing myvoice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is thePSTN. My configuration files are below, I have the latest versionsof Zaptel, Libpri and Asterisk. I am using Polycom’s IP301 and IP430 Phones. I would appreciate help since I have to put this inproduction on Saturday.Well, if that weren't an analog card, I'd say it sounded like clockslip.It could be digital clipping/overdrive; you might check your gains. You have FXS, FXO, and SIP channels, there; which combinations causethe clicking in the transmit audio?Does it happen from FXS to FXO?SIP to FXO?SIP to SIP?How frequently, and how regularly, are the ticks?How loud?How sharp?Can you call someone with audio experience to describe them toyou?(If no one else, feel free to call me; I'm good at this stuff...;-)Cheers,-- jra--Jay R. Ashworth [EMAIL PROTECTED]DesignerBaylink RFC 2100Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USAhttp://baylink.pitas.com +1 727 647 1274That's women for you; you divorce them, and 10 years later,they stop having sex with you.-- Jennifer Crusie; _Fast_Women_ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 wired description and skiping frames
Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw something very weird: Lspci:04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11)It is seen by Ubuntu as an Ethernet board but when I cat /proc/interrupts look: CPU0 0: 949222 IO-APIC-edge timer 1: 8913 IO-APIC-edge i8042 7: 1 IO-APIC-edge parport0 8: 3 IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 410852 IO-APIC-edge i804214: 167803 IO-APIC-edge ide050: 111397 IO-APIC-level eth158: 9467531 IO-APIC-level wctdm24xxp209: 29589 IO-APIC-level libata217: 0 IO-APIC-level ehci_hcd:usb1 225: 0 IO-APIC-level ohci_hcd:usb2233: 2862 IO-APIC-level HDA IntelNMI: 0LOC: 949127ERR: 0MIS: 0The strange thing is that everything seems to be running fast like musiconhold and IVRlike doing fowarding on a tape player... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM2400P
Done it. Didn't work. I am suspicious about some conflict with the Motherboard as it recognizes * as a Ethernet Card -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Thursday, September 21, 2006 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM2400P Try turning off of echotraining. Sometimes it does more harm than good. Matthew Fredrickson On Sep 21, 2006, at 11:28 AM, Robson Ribeiro wrote: Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycoms IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = us defaultzone=us fxsks=1-4 fxsks=5-8 fxoks=17-20 fxoks=21-24 Zapata.conf [channels] language=en context=default ;switchtype=national echocancel=64 echocancelwhenbridged=no echotraining=800 toneduration=200 busydetect=yes signalling = fxs_ks rxgain=5.0 txgain=-10.0 channel = 1-4 channel = 5-8 signalling = fxo_ks channel = 17-20 channel = 21-24 Best Regards, Robson Ribeiro MSN: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 wired description and skiping frames
Hi Tzafrir, i did update the PCI ID and now it shows::04:09.0 Ethernet controller: Digium, Inc. Wildcard TDM2400P (rev 11) What do you get on zttest -v ?here is what i get on zttest -v:Opened pseudo zap interface, measuring accuracy...8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%[2]+ Stopped ./zttest -v Hope it helps me solve the problem i am runnign out of time and ideas.thanks,Robson Ribeiro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 problem isolated with POLYCOM IP301 phones!!!
I managed to isolate the problem. I installed a Sipura ATA 1001 and it does not generates the problem so I figure the issue is with the Polycom phones. Now the issue is to find out whats wrong with the Polycoms. On 9/21/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 21, 2006 at 02:52:03PM -0300, Robson Ribeiro wrote: Erik, the TDM2400 is not in conflict with other interrupts BUT, i saw something very weird: Lspci :04:09.0 Ethernet controller: Digium, Inc.: Unknown device 2400 (rev 11) To get the latest list of PCI IDs descriptions, try:update-pciidslspci -n should tell you the actual numeric IDs. It is seen by Ubuntu as an Ethernet board but when I cat /proc/interrupts look: CPU00: 949222IO-APIC-edgetimer1: 8913IO-APIC-edgei80427:1IO-APIC-edgeparport08:3IO-APIC-edgertc 9:0 IO-APIC-levelacpi 12: 410852IO-APIC-edgei8042 14: 167803IO-APIC-edgeide0 50: 111397 IO-APIC-leveleth1 58:9467531 IO-APIC-levelwctdm24xxp 209:29589 IO-APIC-levellibata 217:0 IO-APIC-levelehci_hcd:usb1 225:0 IO-APIC-levelohci_hcd:usb2 233: 2862 IO-APIC-levelHDA Intel NMI:0 LOC: 949127 ERR:0 MIS:0 The strange thing is that everything seems to be running fast like musiconhold and IVRlike doing fowarding on a tape player... What do you get on zttest -v ?--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ARESKICC DOESN'T make a CALL!!!
Hi Folks,
After going to the paifull steps of installing AreskiCC and finally being able
to access the webinterface, connecting to *, importing rates and setting up
accounts I am not being able to make a CALL: No matter what number i try to
dial I get the same response: The number you have dialed is currently
unavailabel. Please enter thenumber you want to dial starting with 1 for
local and 011 for international. Since I am in Italy there is no such a
numbering but even if I ignore the message and try to dial 3902 (which is in
the ratecard by the way) it tells me the same message. I know it says
destination unreachable but why since it is listed on the ratecard???
*CLI -- Registered 'firefly1' (AUTHENTICATED) at 14.1.172.43:4569
-- Accepting AUTHENTICATED call from 14.1.172.43, requested format = 2,
actual format= 2
-- Executing Answer(IAX2/[EMAIL PROTECTED]/3, ) in new stack
-- Executing Wait(IAX2/[EMAIL PROTECTED]/3, 2) in new stack
-- Executing DeadAGI(IAX2/[EMAIL PROTECTED]/3, areskicc2.php) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc2.php
areskicc2.php: 'agi_request' = 'areskicc2.php'
areskicc2.php: 'agi_channel' = 'IAX2/[EMAIL PROTECTED]/3'
areskicc2.php: 'agi_language' = 'en'
areskicc2.php: 'agi_type' = 'IAX2'
areskicc2.php: 'agi_uniqueid' = '1117858211.0'
areskicc2.php: 'agi_callerid' = 'firefly1 0'
areskicc2.php: 'agi_dnid' = '0290785472'
areskicc2.php: 'agi_rdnis' = 'unknown'
areskicc2.php: 'agi_context' = 'default'
areskicc2.php: 'agi_extension' = '0290785472'
areskicc2.php: 'agi_priority' = '3'
areskicc2.php: 'agi_enhanced' = '0.0'
areskicc2.php: 'agi_accountcode' = '7898389079'
areskicc2.php:
areskicc2.php: ANSWER
areskicc2.php: string(82) firefly1 0 ; IAX2/[EMAIL PROTECTED]/3 ;
1117858211.0 ; 7898389079 ; 0290785472n
areskicc2.php: string(95) SELECT credit, tariff, activated, inuse,
simultaccess FROM cc_card WHERE username='7898389079'n
areskicc2.php: array(1) {n [0]=n array(5) {n[0]=nstring(3)
110n[1]=n string(1) 1n[2]=nstring(1) tn[3]=n
string(1) 1n[4]=nstring(1) 1n }n}n
areskicc2.php: STREAM FILE prepaid-you-have #
areskicc2.php: SAY NUMBER 110 X
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/10' (language 'en')
areskicc2.php: STREAM FILE prepaid-dollars #
areskicc2.php: string(60) UPDATE cc_card SET inuse=inuse+1 WHERE
username='7898389079'n
areskicc2.php: CHANNEL STATUS IAX2/[EMAIL PROTECTED]/3
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: GET DATA prepaid-enter-number-u-calling-1-or-011 1 20
#
-- Playing 'prepaid-enter-number-u-calling-1-or-011' (language 'en')
areskicc2.php: string(36) RES sip_iax_pstndirect_call DTMF : 0n
areskicc2.php: string(22) TRUNK - dnid : 0 (YES)n
areskicc2.php: string(16) YES1 0n
areskicc2.php: GET DATA prepaid-enter-number-u-calling-1-or-011 1 20
#
-- Playing 'prepaid-enter-number-u-calling-1-or-011' (language 'en')
areskicc2.php: string(25) RES DTMF : 39020290785472n
areskicc2.php: string(30) DESTINATION :: 39020290785472n
areskicc2.php: string(34) NEW DESTINATION :: 39020290785472n
areskicc2.php: string(16) RESFINDRATE:: 0n
areskicc2.php: STREAM FILE prepaid-dest-unreachable #
areskicc2.php: CHANNEL STATUS IAX2/[EMAIL PROTECTED]/3
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: GET DATA prepaid-enter-number-u-calling-1-or-011 1 20
#
-- Playing 'prepaid-enter-number-u-calling-1-or-011' (language 'en')
areskicc2.php: string(50) RES sip_iax_pstndirect_call DTMF :
011552122083139n
areskicc2.php: string(36) TRUNK - dnid : 011552122083139 (YES)n
areskicc2.php: string(17) YES15 1n
areskicc2.php: string(31) DESTINATION :: 011552122083139n
areskicc2.php: string(32) NEW DESTINATION :: 552122083139n
areskicc2.php: string(16) RESFINDRATE:: 0n
areskicc2.php: STREAM FILE prepaid-dest-unreachable #
areskicc2.php: CHANNEL STATUS IAX2/[EMAIL PROTECTED]/3
areskicc2.php: result is 6
areskicc2.php: string(20) [CHANNEL STATUS : 6]n
areskicc2.php: GET DATA prepaid-enter-number-u-calling-1-or-011 1 20
#
-- Playing 'prepaid-enter-number-u-calling-1-or-011' (language 'en')
== Spawn extension (default, 0290785472, 3) exited non-zero on
'IAX2/[EMAIL PROTECTED]/3'
-- Hungup 'IAX2/[EMAIL PROTECTED]/3'
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pri doesn't accept Zap/g2 to call
I have a Sangoma Card with two PRIs. They are both
configured in Zaptel and Zapata;
In Zapata I have them separated in Group 1 and 2 but if I
make a call and specify Zap/g2 it doesnt go when calling Channels :
HERE IS what I get:
Accepting AUTHENTICATED call from x.x.x.x
requested format = speex,
requested prefs =
(),
actual format =
gsm,
host prefs =
(ilbc|gsm),
priority = mine
-- Executing
Dial(IAX2/[EMAIL PROTECTED], Zap/g2/3337885836|100|T)
in new stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called g2/3337885836
-- Channel 0/1, span 2 got hangup request
-- Hungup 'Zap/32-1'
== No one is available to answer at this time
(1:0/0/0)
-- Executing Hangup(IAX2/[EMAIL PROTECTED],
) in new stack
== Spawn extension (default, 3337885836, 2) exited
non-zero on 'IAX2/[EMAIL PROTECTED]'
-- Hungup 'IAX2/[EMAIL PROTECTED]'
Zapata.conf
[globals]
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
[default]
exten = _X.,1,Dial(${PRITRUNK2}/${EXTEN},100,T)
exten = _X.,2,Hangup
[firefly1]
exten = _X.,1,Dial(${PRITRUNK1}/${EXTEN},100,T)
exten = _X.,2,Hangup
Zaptel.conf:
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-61
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI doesn't call cellphones
Hi all, I am using a Sangoma with two PRIs. As far as land phones, the calls are fine but it refuses all cellphone calls: My configuration in Zaptel is span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-61 and on Zapata.conf: [channels] language=it context=default switchtype=national signalling=pri_cpe echocancel=yes group=1 callgroup=1 pickupgroup=1 when I call a cellphone I get the following error from Pri Span Debug: login as: root Authenticating with public key rsa-key-20050520 Passphrase for key rsa-key-20050520: Wrong passphrase Authenticating with public key rsa-key-20050520 Passphrase for key rsa-key-20050520: Wrong passphrase Authenticating with public key rsa-key-20050520 Passphrase for key rsa-key-20050520: Last login: Sat May 21 13:16:10 2005 from 212.102.34.109 Linux 2.4.29. [EMAIL PROTECTED]:~# cd /etc/asterisk [EMAIL PROTECTED]:/etc/asterisk# joe zapata.conf Processing '/etc/joe/joerc'...done Processing '/etc/joe/joerc'...done I zapata.conf (Modified) Row 15 Col 1 1:56 Ctrl-K H for help language=it context=default switchtype=national signalling=pri_cpe echocancel=yes group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-61 [channels] language=it context=default switchtype=national signalling=pri_cpe echocancel=yes group=1 callgroup=1 Verbosity is at least 7 -- Remote UNIX connection -- Executing Dial(SIP/200-4f96, Zap/g1/3337885836|100|T) in new stack -- Making new call for cr 32778 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 10/0xA) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] 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 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 06 b1 55 73 65 72 31] Display (len= 6) Charset: 31 [ User1 ] [6c 05 21 81 32 30 30] 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 passed network screening (1) '200' ] [70 0b a1 33 33 33 37 38 38 35 38 33 36] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3337885836' ] -- Called g1/3337885836 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 10/0xA) (Terminator) Message type: STATUS (125) [08 03 82 e3 28] Cause (len= 5) [ 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 1: 28 (40, Display IE) [14 01 01] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 10/0xA) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] 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 ] [1e 02 82 88] Progress Indicator (len= 4) [ 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 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 10/0xA) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ 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 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) -- Channel 0/1, span 1 got hangup request NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 10/0xA) (Originator) Message
[Asterisk-Users] IVR/Voicemail, No Sound from Asterisk
Hi all, I am having a problem with a recent installed *. The IVR, voicemail internal greeting sounds dont play!. I see on the CLI interface that it is playing but I cant hear anything. I have the following configuration on the asterisk. - Current Asterisk CVS - A TDM400 with 4 FXOs - A FRITZ ISDN using CAPI - Linux Debian 2.4.27 Thanks. Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AreskiCC
Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it andobviously t doesnt work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load /areskicc/Public/index.php it refuses my username and passwork (AUTHENTICATION REFUSED, please check your login/password! ) which I guess is the same as the one I configured on defines.php right?) and after I reinsert it I get the error: Method Not Allowed. The requested method POST is not allowed for the URL /areskicc/Public/index2.php. In any case, does anybody know of any better instructions on how to install and configure AreskiCC? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AreskiCC
Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it andobviously t doesnt work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load /areskicc/Public/index.php it refuses my username and passwork (AUTHENTICATION REFUSED, please check your login/password! ) which I guess is the same as the one I configured on defines.php right?) and after I reinsert it I get the error: Method Not Allowed. The requested method POST is not allowed for the URL /areskicc/Public/index2.php. In any case, does anybody know of any better instructions on how to install and configure AreskiCC? Thanks, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Compilation Error
Hi, When trying to compile ASTCC i am getting the following error: [EMAIL PROTECTED]:/usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.6/i486-linux /usr/lib/perl5/5.8.6 /usr/lib/perl5/site_perl/5.8.6/i486-linux /usr/lib/perl5/site_perl/5.8.6 /usr/lib/perl5/site_perl .) at ./astcc.agi line 47. BEGIN failed--compilation aborted at ./astcc.agi line 47. make: *** [install] Error 2 Anyone can help please? Thanks, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold can' t hear it!
I have the current version og mpg running. But i am geeting the same problem even with the ringing tone. It seems to disappear sometimes make[1]: Entering directory `/usr/src/mpg123-0.59r' make[2]: Entering directory `/usr/src/mpg123-0.59r' make[2]: `mpg123' is up to date. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold can' t hear it!
Hi folks, I am having a problem with MusicOnHold. Right now I have the following configuration: Default = mp3:/var/lib/asterisk/mohmp3 The problem is that I can't hear the music or sometimes the music seems to skip like a scratched record... Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial While on IVR
Title: Dial While on IVR While the call is going into the IVR how can I Dial an extension and get immediately connected interrupting the IVR? Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Error
Anyone has any idea what does this error means when executing an IAX2 call? Apr 22 11:50:19 WARNING[9124]: Received mini frame before first full voice frame The called party can hear but the calling, no. Is this a fine tunning into iax.conf? Thanks, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Error
Guys, every now and then this error comes up. I can't heat the party i Call. It doesn't seem normal: Apr 22 17:57:20 WARNING[9124]: chan_iax2.c:6006 socket_read: Received mini frame before first full voice frame Apr 22 17:57:20 WARNING[9124]: chan_iax2.c:6006 socket_read: Received mini frame before first full voice frame Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO GW Dial in/out syntax
Title: FXO GW Dial in/out syntax Hi all, I have a non-branded FXO Gateway connected to 4 analog lines at the office. The situation is that I figured out how to make it dial in with the following entries: In sip.conf: [4003] username=4003 fromuser=4003 dtmfmode=rfc2833 type=friend secret=4003 host=dynamic context=gw_fxo and then, in extensions: [gw_fxo] exten = _X.,1,Goto(mainmenu,s,1) ; so that it can answer any of the 4 ports and send it to IVR. But to dial out, I am suffering to understand: In extensions I put at the outgoing context: Exten = _9x,1,Dial,SIP/[EMAIL PROTECTED]/$(EXTEN:1) Doesnt work. Also i tried registering the GW into SIP so it looked like an account but it doesnt access registration. Thanks in advance for the help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home ISDN BRI
Frtiz is a nightmare although it is cheap and I have seen it working. I have been trying to install it for some days without success but one thing is for sure: you have to use the right Kernel (they are available for 2.4.20 and 2.6something). There are no clear instructions to install it, other than those found at: http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install that I have followed but it seems the code is not correct. I will keep you posted because I was trying to make it work on kernel 2.4.29 and nothing. I am installing another box with the kernel 2.4.20 (ftp://ftp.kernel.org). Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling 2nd AVM Fritz
Title: [Asterisk-Users] Problem compiling 2nd AVM Fritz I am having the exact problem. I managed to get to only 1 error by making sure the paths were correct. But yesterday 11PM the achine froze. Only this morning i will find out whats wrong. Robson Shane Dalgleish asterisk at tragicflirt.com Wed Apr 6 06:38:46 CDT 2005 I am adding an extra AVM Fritz card to an existing setup.. I have followed instructions from http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO I did a make clean before I started, made all the changes as specified on the quiss.org site, and the results are shown below.. [root at sip fritz]# make (cd src.drv; make CARD=f2pci) make[1]: Entering directory `/usr/src/fritz/src.drv' cc -c -DMODULE -DMODVERSIONS -D__KERNEL__ -DNDEBUG \ -march=i686 -O2 -Wall -I /lib/modules/`uname -r`/build/include \ main.c -o main.o cc: cannot specify -o with -c or -S and multiple compilations make[1]: *** [main.o] Error 1 make[1]: Leaving directory `/usr/src/fritz/src.drv' make: *** [drv] Error 2 Any thoughts anyone? Shane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fritz Card going Crazy to make it compile
I have reviewed everything, notes etc but I can't get the fcpci.o to show up in the src.drv or anytother directory. Here is what I am getting when I do make: [EMAIL PROTECTED]:/usr/src/fritz/src.drv# make cc -c -DMODULE -DMODVERSIONS -D__KERNEL__ -DNDEBUG -D -DTARGET=\\ -march=i386 -O2 -Wall -I /usr/src/linux/include -include /usr/src/linux/include/linux/modversions.h main.c -o main.o In file included from /usr/src/linux/include/linux/spinlock.h:6, from /usr/src/linux/include/linux/wait.h:16, from /usr/src/linux/include/linux/fs.h:12, from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/asm/uaccess.h:8, from main.c:28: /usr/src/linux/include/asm/system.h: In function `__set_64bit_var': /usr/src/linux/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules /usr/src/linux/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules In file included from tools.h:30, from main.c:48: defs.h: At top level: defs.h:89: error: redefinition of `irqreturn_t' /usr/src/linux/include/linux/interrupt.h:16: error: `irqreturn_t' previously declared here main.c:60: error: parse error before PRODUCT_LOGO main.c: In function `fritz_init': main.c:140: error: `PRODUCT_LOGO' undeclared (first use in this function) main.c:140: error: (Each undeclared identifier is reported only once main.c:140: error: for each function it appears in.) main.c:140: error: parse error before string constant make: *** [main.o] Error 1 [EMAIL PROTECTED]:/usr/src/fritz/src.drv# At this point I am stuck. Or if anybody can send me a compiled fcpci.o for Kernel 2.4.29 (Slackware) I appreciate. Robson attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Fritz and TDM400
Is it possible to have on the same machine an ISDN Fritz Card and a TDM400 with two FXO ports? If so, is there any place I can find instructions to configure it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Fritz and TDM400
Thanks for your reply, my doubt rest on the fact that there are two ways of configuring it: One using the Bristuff from Junghanns and the other using CAPI. Is there any major difference/advantages to one or the other? p.s. I cant find instructions on how to configure bristuff besides what comes with the package. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel and Fritz Card
Does anyone has instructions on how to install the Fritz PCI Card with Zaptel? There is no clear instructions in Junghanns.net nor on the Fritz Card ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel and Fritz Card
Hi Oliver, I am trying to install only the Fritz Card. But according to the instructions on: http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install it doesnt work. The directories, even the changes that they suggest on the makefile are not there!! I am really disappointed I have been on this for hours!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Fritz and TDM400
Title: [Asterisk-Users] ISDN Fritz and TDM400 Damian, thanks for the support. Until now I have had no success with all the kernel compilations, ISDN CAPI Utilities etc. I am almost there but it sucks. Wouldnt be a better way just to simply do it withouth having to spend so much time with it? SO far I am having trouble compiling the FRITZ PCI stuff and getting fcpci.o to show upI just froze the remote instance so I guess only tomorrow now.;( I will keep you posted of my progress and thanks for the drivers, when I get there I might need them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Micronet 128K TA Card
Does anybody has instructions on how to configure the Micronet ISDN 128K card with Asterisk? Thanks, Robson Ribeiro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Micronet 128K TA Card
Do you mean the same as the AVM Fritz PCI Card? Does it uses the same CAPI and other drivers? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music Answer while waiting
Hi, If I want a user to, while waiting for a transfer after responding to an IVR, to listen to music instead of a ring sound, what is the change should i do in extensions.conf? Is it on the IVR menu or on the optional extension Txs, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM04B doesn't hang up after Voicemail
Hello all,
I am having a serious problem installing my * with a TDM04B. I made everything
work, call are coming in and going out including using a GSM Box in channel
Zap/2-1. I did setup voicemail like this on extensions.conf:
[incoming]
exten = s,1,Dial(SIP/2246,20)
exten = s,2,Wait,2
exten = s,3,Voicemail(u${ME})
exten = s,4,Hangup
exten = s,102,Wait,2
exten = s,103,Voicemail(b${ME})
exten = s,104,Hangup
After the call is finished if the user doesn't press # the line hangs forever.
Unfortunately I found it out after i did a zap show... 26 minutes after
the call ended :(. I looked into the threads but no answer seems to resolve
the problem (maxthreashhold or maxsilence and there is even a patch to one of
the voicemail files which i have no idea how to implement). The other strange
thing it is happening is that after i hang up the call from the phone if the
outside caller hasn't hang up it recreates the Zap channel and rings it
again.any clues please?
Thanks for the help.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI and SIP PHones in Asterisk
Does anybody has a link for a step by step explanation on how dows MWI works in Asterisk with a SIP phone? I hacve added the mailbox line in SIP.conf but i got nothing :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't Dial Out with TDM04B
Hi and thank you.
I am a beginer trying to install my first TDM04B.
I am able to receive call with the card using:
[incoming]
exten = s,1,Dial(SIP/robgol,20,tr)
on my extensions
but, with
[outgoing]
exten = _0X.,1,Zap/1/${EXTEN}
I cant send them out. I am getting the following error:
Mar 26 23:37:07 WARNING[5189]: pbx.c:1291
pbx_extension_helper: No application 'Zap/1/${EXTEN:1}' for
extension (default, 00290785472, 1)
== Spawn extension (default, 00290785472, 1) exited
non-zero on 'SIP/robgol-bf04'
thanks in advance for the help.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't Dial Out with TDM04B
Hi and thank you.
I am a beginer trying to install my first TDM04B.
I am able to receive call with the card using:
[incoming]
exten = s,1,Dial(SIP/robgol,20,tr)
on my extensions
but, with
[outgoing]
exten = _0X.,1,Zap/1/${EXTEN}
I cant send them out. I am getting the following error:
Mar 26 23:37:07 WARNING[5189]: pbx.c:1291
pbx_extension_helper: No application 'Zap/1/${EXTEN:1}' for
extension (default, 00290785472, 1)
== Spawn extension (default, 00290785472, 1) exited
non-zero on 'SIP/robgol-bf04'
thanks in advance for the help.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't Dial Out with TDM04B
Daniel, i found the problem. Actually besides a mistype error on the extensions.conf. It was all about USB conflict. I went into the BIOS and reserverd IRQ's 11,12,13 and 14 (just to be safe) and it worked. After i found the mistype i could only send and receive calls on the first channel although they all appeared. So this might be a cointribution or it was already a fact and i didn't know about. thanks anyway. made my day. Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
