Re: [asterisk-users] Asterisk and T38 ?
Tobias Wolf a écrit : Noc Phibee schrieb: Hi i read the list and see a lot of personn say T38 it's not possible with asterisk and other says that he use T38 with asterisk ?? i don't understand ;=) Well, if i understand it correctly then Asterisk currently only supports T.38-Passthrough, which means, you have to have to T.38 capable Endpoints which can communicate with an Asterisk in the Media Path. But you cannot terminate an T.38 Call on an Asterisk Server (say receiving an Fax with an Asterisk and saving the Fax as an TIFF on the server). Anyone feel free to correct me if i am wrong ;) Cheers, Tobias Thanks for your answer ;=) We don't have a solution for use a codec without comrpession for supply a line at a Fax and at a modem ? Modem/Fax with VoIP never work ? ___ --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] Asterisk and T38 ?
Hi i read the list and see a lot of personn say T38 it's not possible with asterisk and other says that he use T38 with asterisk ?? i don't understand ;=) He have a solution (commercial or free) to add T38 ? I have : Fax Machine -- Linksys PAPT -- Asterisk === IAX2 on Sdsl === Asterisk -- PSTN E1 what is the solution ? thanks ___ --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] Update Asterisk 1.2.12 to 1.4.1 ?
Hi i have a big change or bproblems to update a asterisk 1.2.12 server to asterisk 1.4.1 ? Thanks bye ___ --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] Problems Asterisk with Digium TDM400 card = he don't see the disconnect
Hi i have a big problems with my asterisk .. i use a Digium TDM400P for connect a analog line. And not all time (i don't know why) he don't see the end of the call and anyone can call me (occuped) For that's work, i am disconnect the phone cable and it's good anyone have a idea ? bye ___ --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] Cisco Router for supply a connection from PABX to Asterisk ?
Hi anyone know if they have a solution in Cisco for: 1- Connect old PABX (with BRI or PRI) to a cisco router 2- Connect this cisco router in SIP to a Asterisk Server I am search if cisco can this and what is the modele for this Thanks ;=) ___ --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] Problems with GXP2000 and Asterisk = Call pickup and Voicemail
Hi thanks for your answer, for dtmfmode, all sip account have dtmfmode=rfc2833 ;=) that's don't change bye Gordon Henderson a écrit : On Fri, 9 Feb 2007, Noc Phibee wrote: Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me [EMAIL PROTECTED] not found .. in features.conf, i have: [general] parkext = 700parkpos = 701-720 context = parkedcalls pickupexten = *8 I'm under the impression that *8 picks up any ringing phone in the same group... Not sure why youre dialling an extension number after it... I may be wrong though - I've never used it! 2- When i want access to the voice server, he never understand my password ... but with a softphone that's work's Anyone have this problems too ? I'd guess that asterisk isn't hearing the tones of the password? Start with putting dtmfmode=rfc2833 in your sip.conf file, and making that setting on the GPX2000 phone itself (on the account page) Gordon ___ --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] Problems with GXP2000 and Asterisk = Call pickup and Voicemail
Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me [EMAIL PROTECTED] not found .. in features.conf, i have: [general] parkext = 700 parkpos = 701-720 context = parkedcalls pickupexten = *8 2- When i want access to the voice server, he never understand my password ... but with a softphone that's work's Anyone have this problems too ? Thanks bye ___ --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] Use Digium TE110P Single T1 / E1 PCI Interface Card for connect a old PABX ?
Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks ___ --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] Asterisk IAX and Shorewall QoS ?
Hi anyone have a sample of shorewall configuration for add a TC/QoS on IAX2 traffic ? Thanks for your help ___ --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] Grandstream GXP2000 and Interception of call ?
Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone Thanks bye ___ --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] 2 Questions: Answer with music don't work and Voicemail direct access ?
Hi Stefan,
Thanks for your answer, but it's a error of me in cut, the goto are good:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Internal]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten = 100,4,Dial(SIP/220SIP/221,30)
exten = 100,5,Hangup
exten = 200,1,Ringing
exten = 200,2,Wait,1
exten = 200,3,Answer
exten = 200,4,Dial(SIP/221,25,tm)
exten = 200,5,Hangup
;=)
Stefan Wintermeyer a écrit :
Hi,
Am 17.01.2007 um 15:07 schrieb Noc Phibee:
Problems with Answer+Music
my extension:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Phibee]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten = 100,4,Dial(SIP/201SIP/200,30)
exten = 100,5,Hangup
exten = 200,1,Ringing
exten = 200,2,Wait,1
exten = 200,3,Answer
exten = 200,4,Dial(SIP/200,25,tm)
exten = 200,5,Hangup
With this extension, when a incoming call are received :
If my customer have call 081120, that's answer and Ring
If my customer have call 081121, he have a answer, he have a
music
I don't know why the 081120 don't have the music for wait that i
am answer ...
I guess you simply did a mistake in the Goto. It points to the
C-Internal context but you want to jump to C-Phibee.
It's possible to put into the extension, for access to the VoiceMail,
the extension of the caller ?
exten = 500,1,VoiceMailMain(@Home)
Actually, when i call the 500, he want know my mailbox ID and after
password ...
if i call with the post 200, it's possible to access direclty at the
password ?
Yes:
exten = 500,1,VoiceMailMain(${CALLERID(num)[EMAIL PROTECTED])
But I am not sure if you really want to use @Home here. But that
depends on you voicemail.conf
BTW: With exten = 500,1,VoiceMailMain(${CALLERID(num)[EMAIL PROTECTED],s) you
can even skip the password question.
Stefan
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
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[asterisk-users] 2 Questions: Answer with music don't work and Voicemail direct access ?
Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten = _81120,1,Goto(C-Internal,100,1) exten = _81121,1,Goto(C-Internal,200,1) [C-Phibee] exten = 100,1,Ringing exten = 100,2,Wait,1 exten = 100,3,Answer exten = 100,4,Dial(SIP/201SIP/200,30) exten = 100,5,Hangup exten = 200,1,Ringing exten = 200,2,Wait,1 exten = 200,3,Answer exten = 200,4,Dial(SIP/200,25,tm) exten = 200,5,Hangup With this extension, when a incoming call are received : If my customer have call 081120, that's answer and Ring If my customer have call 081121, he have a answer, he have a music I don't know why the 081120 don't have the music for wait that i am answer ... Second Question: It's possible to put into the extension, for access to the VoiceMail, the extension of the caller ? exten = 500,1,VoiceMailMain(@Home) Actually, when i call the 500, he want know my mailbox ID and after password ... if i call with the post 200, it's possible to access direclty at the password ? Thanks bye ___ --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] Secure a Asterisk Server ?
Hi actually, i have only one Asterisk Server ;=) Anyone know a how to for create a seconde asterisk in Backup for hight availability ? Thanks bye ___ --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] Create a group of SIP acoount for outgoing calls ?
Hi
actually, for call i use ZAP Channels on a E1 and SIP Account on a VoIP
provider ...
in Zap, we use group and we have:
exten = _1.,2,Dial(Zap/r1/${EXTEN:1},50,rt)
exten = _1.,3,Hangup
r1= he change of channels at all calls channel group 1
It's possible to create a group of SIP Account and use same r1
for outgoing calls ?
r2= he don't use the same account into sip group 2
Thanks bye
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[asterisk-users] System() and Trysystem() in extensions.conf = get the result ?
Hi, if i use System() or TrySystem() into my extensions.conf for execute a external command, can i get and put the result of the command into a variable ? Thanks bye ___ --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] Digium TE405P with French E1 = Red Alert
Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes immediate=no amaflags=documentation musiconhold=default group=1 callgroup=1 pickupgroup=1 channel = 1-15 channel = 17-31 a ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. but with all test, i have a red alert: ipbx*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1RED0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ipbx*CLI i use a crossover cable: 1=4 2=5 4=1 5=2 to my PRI supplier My syslog: Dec 18 12:46:39 ipbx kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Dec 18 12:46:39 ipbx zaptel: Loading zaptel framework: succeeded Dec 18 12:46:39 ipbx kernel: ACPI: PCI Interrupt :00:05.0[A] - GSI 24 (level, low) - IRQ 24 Dec 18 12:46:39 ipbx kernel: Found TE4XXP at base address febffc00, remapped to f8afec00 Dec 18 12:46:39 ipbx kernel: TE4XXP version c01a016a, burst OFF, slip debug: OFF Dec 18 12:46:39 ipbx kernel: FALC version: 0005, Board ID: 00 Dec 18 12:46:39 ipbx kernel: Reg 0: 0x186ec400 Dec 18 12:46:39 ipbx kernel: Reg 1: 0x186ec000 Dec 18 12:46:39 ipbx kernel: Reg 2: 0x Dec 18 12:46:39 ipbx kernel: Reg 3: 0x Dec 18 12:46:39 ipbx kernel: Reg 4: 0x Dec 18 12:46:39 ipbx kernel: Reg 5: 0x Dec 18 12:46:39 ipbx kernel: Reg 6: 0xc01a016a Dec 18 12:46:39 ipbx kernel: Reg 7: 0x1000 Dec 18 12:46:39 ipbx kernel: Reg 8: 0x010200ff Dec 18 12:46:39 ipbx kernel: Reg 9: 0x00fd Dec 18 12:46:39 ipbx kernel: Reg 10: 0x Dec 18 12:46:39 ipbx kernel: TE4XXP: Launching card: 0 Dec 18 12:46:39 ipbx kernel: TE4XXP: Setting up global serial parameters Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 0 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 1 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 2 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 3 Dec 18 12:46:39 ipbx kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) Dec 18 12:46:40 ipbx kernel: About to enter spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configuring span 1 Dec 18 12:46:40 ipbx kernel: Done with spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/1! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/2! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/3! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 4 (TE4/0/1/4) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel
Re: [asterisk-users] Digium TE405P with French E1 = Red Alert
Hi it's Colt-Telecom. you have a TE405P ? bye pixiesfr a écrit : Hi what is your operator? I have some pb on orange... thx Noc Phibee a écrit : Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes immediate=no amaflags=documentation musiconhold=default group=1 callgroup=1 pickupgroup=1 channel = 1-15 channel = 17-31 a ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. but with all test, i have a red alert: ipbx*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1RED0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ipbx*CLI i use a crossover cable: 1=4 2=5 4=1 5=2 to my PRI supplier My syslog: Dec 18 12:46:39 ipbx kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Dec 18 12:46:39 ipbx zaptel: Loading zaptel framework: succeeded Dec 18 12:46:39 ipbx kernel: ACPI: PCI Interrupt :00:05.0[A] - GSI 24 (level, low) - IRQ 24 Dec 18 12:46:39 ipbx kernel: Found TE4XXP at base address febffc00, remapped to f8afec00 Dec 18 12:46:39 ipbx kernel: TE4XXP version c01a016a, burst OFF, slip debug: OFF Dec 18 12:46:39 ipbx kernel: FALC version: 0005, Board ID: 00 Dec 18 12:46:39 ipbx kernel: Reg 0: 0x186ec400 Dec 18 12:46:39 ipbx kernel: Reg 1: 0x186ec000 Dec 18 12:46:39 ipbx kernel: Reg 2: 0x Dec 18 12:46:39 ipbx kernel: Reg 3: 0x Dec 18 12:46:39 ipbx kernel: Reg 4: 0x Dec 18 12:46:39 ipbx kernel: Reg 5: 0x Dec 18 12:46:39 ipbx kernel: Reg 6: 0xc01a016a Dec 18 12:46:39 ipbx kernel: Reg 7: 0x1000 Dec 18 12:46:39 ipbx kernel: Reg 8: 0x010200ff Dec 18 12:46:39 ipbx kernel: Reg 9: 0x00fd Dec 18 12:46:39 ipbx kernel: Reg 10: 0x Dec 18 12:46:39 ipbx kernel: TE4XXP: Launching card: 0 Dec 18 12:46:39 ipbx kernel: TE4XXP: Setting up global serial parameters Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 0 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 1 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 2 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 3 Dec 18 12:46:39 ipbx kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) Dec 18 12:46:40 ipbx kernel: About to enter spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configuring span 1 Dec 18 12:46:40 ipbx kernel: Done with spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/1! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/2! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 Dec 18 12:46:40 ipbx
Re: [asterisk-users] Multi Operator
Hi I don't see a answer to this question ;=) i am search this solution too .. Thanks bye Jea philippe a écrit : Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1= Dial SIP/phone1 Call 2= Dial SIP/phone2 Call 3= Dial SIP/phone1 ... If you have an sample please let me know ___ --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] send fax by Iaxmodem ?
Hi i use now iaxmodem for receive fax and that's work very good with Hylafax ;=) Do you know if we can sent fax using iaxmodem and Hylafax ? when i test: déc 13 13:47:21.12: [13725]: SESSION BEGIN 00014 330426690268 déc 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0 déc 13 13:47:21.12: [13725]: SEND FAX: JOB 2 DEST 0426690268 COMMID 00014 DEVICE '/dev/iaxmodem1' FROM 'localtest' USER test déc 13 13:47:21.12: [13725]: STATE CHANGE: RUNNING - SENDING déc 13 13:47:21.12: [13725]: -- [12:AT+FCLASS=r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled déc 13 13:47:21.12: [13725]: DIAL 0426690268 déc 13 13:47:21.12: [13725]: -- [15:ATDT0426690268\r] déc 13 13:47:21.12: [13725]: -- [11:NO DIALTONE] déc 13 13:47:21.12: [13725]: SEND FAILED: JOB 2 DEST 0426690268 ERR No local dialtone déc 13 13:47:21.12: [13725]: -- [5:ATH0\r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set DTR OFF déc 13 13:47:21.12: [13725]: MODEM set baud rate: 0 baud (flow control unchanged) déc 13 13:47:21.12: [13725]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) déc 13 13:47:21.12: [13725]: SESSION END Thanks bye ___ --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] Asterisk and Fax How To
Hi i have a asterisk server with a Digium 4xE1 card connected to my local operator. I am search a How to for : - Add a Mail to Fax server - Add a Fax to Mail Server thanks bye ___ --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] Anonymous clid ?
Hi for put a anonymous clid on a out line sip, what is the config ? thanks bye ___ --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] Asterisk and TDM400P ?
Thanks Giogio, but no i don't have this module bye Giorgio Incantalupo a écrit : Hi Noc, I had similar problem. Check If you have netjetpci module and try to delete it...this solved my problem. Giorgio Incantalupo Noc Phibee wrote: Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 24 06:13:43 gw kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Nov 24 06:13:44 gw kernel: Registered tone zone 2 (France) Nov 24 06:13:44 gw zaptel: Running ztcfg: succeeded and my problems are whit all sample that i have, asterisk don't restart and put me: Nov 24 06:15:17 NOTICE[2943] cdr.c: CDR simple logging enabled. Nov 24 06:15:17 WARNING[2943] chan_zap.c: Unable to specify channel 3: No such device Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to open channel 3: No such device here = 0, tmp-channel = 3, channel = 3 Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to register channel '3' Nov 24 06:15:17 WARNING[2943] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 06:15:17 WARNING[2943] loader.c: Loading module chan_zap.so failed! for all channel (i have tested from 1 to 5) my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 where is my errors ? Thanks for your help ___ --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] Asterisk and TDM400P ?
thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Leo Ann Boon a écrit : Noc Phibee wrote: Thanks Giogio, but no i don't have this module bye Check your zapata.conf. Your signalling and channel settings are wrong for FXO module. signalling=fxs_ls channel= 4 FXO module use fxs signalling, FXS module use fxo signalling. Leo. ___ --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] Asterisk and TDM400P ?
Leo Ann Boon a écrit : Noc Phibee wrote: thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Can you check if your /dev/zap directory is created correctly? On my machine with a TDM400P with 2xFXS and 2xFXO. [EMAIL PROTECTED] ~]$ ls /dev/zap/ 1 2 3 4 channel ctl pseudo time If you don't see anything then you'll have to check if your security setting is prevent access to /dev/zap. Leo Yes i have ;=) [EMAIL PROTECTED] zap]# ll total 0 crw-rw 1 asterisk asterisk 196, 1 nov 24 06:29 1 crw-rw 1 asterisk asterisk 196, 2 nov 24 06:29 2 crw-rw 1 asterisk asterisk 196, 3 nov 24 06:29 3 crw-rw 1 asterisk asterisk 196, 4 nov 24 06:29 4 crw-rw 1 asterisk asterisk 196, 254 nov 24 06:29 channel crw-rw 1 asterisk asterisk 196, 0 nov 24 06:29 ctl crw-rw 1 asterisk asterisk 196, 255 nov 24 06:29 pseudo crw-rw 1 asterisk asterisk 196, 253 nov 24 06:29 timer ___ --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] Asterisk and TDM400P ?
Pranav Peshwe a écrit : Hi, Check your /etc/zaptel.conf and ensure that it has the right kind of signalling set for the same channel number as that in you zapata.conf. do : cat /proc/zaptel/1 and it should show channels and the effective signalling settings for them. If signalling does not appear here,it means that, it is not configured properly, and loading chan_zap would fail. My first and fourth channels are configured as fxsks and the output i get is : #cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 FXSKS HTH :) Regards, Pranav Thanks for your help, a cat: [EMAIL PROTECTED] zap]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 in my zaptel.conf, i have only: loadzone=fr defaultzone=fr fxsks=1 and zapata.conf: [trunkgroups] [channels] context=default signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default callerid=Filtrinov0477530573 channel = 5 if i understand, my error are channel ? In zaptel.conf, its fxsks=1 and in zapata.conf it's channel = 0 no ? ___ --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] Asterisk and TDM400P ?
Tzafrir Cohen a écrit : * Use genzaptelconf from xpp/utils/genzaptelconf to save you from this guesswork. Hi, thanks ;=) with genzaptelconf, now that's works ... correct channel are put into zaptel.conf and zapata.conf small question if you know the TDM400P: if the fxo module are at the slot 4, the RJ11 connector are the number 4 ? a show channels done: gw*CLI zap show channels Chan Extension Context Language MusicOnHold pseudointerne 4interne gw*CLI gw*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 now, i can add to my extension ZAP/4 ;=) for see if the card answer, what is the process ? very very thanks at all for this result ___ --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] Asterisk and TDM400P ?
Hi, i receive a call on my analog line but my asterisk don't answer ;=) do you know if they hae a solution for know if the card see the call ? for see if it's not my cable don't work .. thanks bye ___ --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] Asterisk and TDM400P ?
Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 24 06:13:43 gw kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Nov 24 06:13:44 gw kernel: Registered tone zone 2 (France) Nov 24 06:13:44 gw zaptel: Running ztcfg: succeeded and my problems are whit all sample that i have, asterisk don't restart and put me: Nov 24 06:15:17 NOTICE[2943] cdr.c: CDR simple logging enabled. Nov 24 06:15:17 WARNING[2943] chan_zap.c: Unable to specify channel 3: No such device Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to open channel 3: No such device here = 0, tmp-channel = 3, channel = 3 Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to register channel '3' Nov 24 06:15:17 WARNING[2943] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 06:15:17 WARNING[2943] loader.c: Loading module chan_zap.so failed! for all channel (i have tested from 1 to 5) my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 where is my errors ? Thanks for your help ___ --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] Question on CDR Database
Hi I have a small question on CDR Database: It's used by billing software no ? he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended thanks bye ___ --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] If of external small box supply fxs Isdn and E1 ?
Hi anyone know a list of external hardware supported by asterisk for connect old Pbx to VoIP line ? For supply Isdn BRI and PRI to my clients thanks ___ --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] AdvancedVoIP Billing ?
Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ --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] AdvancedVoIP Billing ?
Yes ;=) but a2billing it's for calling card ;) Al Bochter a écrit : Did you look at a2billing? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (VOIP PBX) 1-866-638-1254 (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Noc Phibee wrote: Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ ___ --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] AdvancedVoIP Billing ?
Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html ___ --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] Trunk outcall line ?
Hi
actually, for out call, i use :
exten = _0.,1,Dial(SIP/out-l1/${EXTEN:1},50,rt)
exten = _0.,2,Dial(SIP/out-l2/${EXTEN:1},50,rt)
exten = _0.,3,Dial(SIP/out-l3/${EXTEN:1},50,rt)
exten = _0.,4,Hangup
can you say me with this config, if the first user call and use out-l1
the second user use automatiquely out-l2 (and out-l3 when l1 and l2 are
used) ?
if i want add a turn line for use all lines and not only the out-l1,
what is the best
config (i don't use web interface for config)
thanks bye
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[asterisk-users] Asterisk and FXO Digium Card for Analog line
Hi For add a analog line to my asterisk, i want add a Dgium Fxo card. but i want know a small information: The quality of the call are good or not with this type of card ? Thanks for your returns ___ --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] Grandstream HandyTone-488 with Asterisk ?
Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye ___ --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] Billing Solution ?
Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye ___ --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] Billing Solution ?
Thanks all for your answer ;=) i start test this week a2billing Noc Phibee a écrit : Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye ___ ___ --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] Fxo box for asterisk ?
Hi do you know if they have external Box (not internal card) for connect Analog Line and Pri/Isdn to asterisk for incomming and outgoing calls ... Thanks ___ --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] Bri Card for Asterisk ?
Hi a small question: what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ? Thanks bye ___ --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] Asterisk and Maximum retries exceeded
Hi today, i have a big problems with my asterisk ... when i want call i have this error : Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1243 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. srv1*CLI for all phone and i don't have change my configuration anyone have a idea of the problems ? Thanks bye ___ --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] Asterisk and Maximum retries exceeded
anyone know this error ?? Noc Phibee a écrit : Hi today, i have a big problems with my asterisk ... when i want call i have this error : Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1243 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. srv1*CLI for all phone and i don't have change my configuration anyone have a idea of the problems ? Thanks bye ___ --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] Asterisk and NAT ?
Hi I am search a small information - i use Asterisk on official IP without Nat - My first VoIP phone are a Thomson 2030 on a NAT Network. That's work very good. But now, i want add a second phone, a Linksys SPA-941 on the same network of the Thomson 2030 ... My problems that i don't see a solution into asterisk or on my firewall for that's work. When i call to the thomson, that's work, when i call to the linksys that's don't ring ... On my asterisk i have put : 200= thomson 202= linksys [200] port=5060 username=200 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne qualify=yes nat=route dtmfmode=rfc2833 language=fr [202] port=5070 username=202 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=route dtmfmode=rfc2833 language=fr on my firewall, i have put a forward of port 5060 to thomson and 5070 to linksys in UDP and TCP. On linksys i can call but not receive call on thomson i can call and receive without problems Thanks for your help ___ --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] Asterisk and NAT ?
yusuf a écrit : Hi, you dont have to/should'nt be using different SIP ports for each phone. Its completely not needed. Also, you dont have/need to port forward. Just open ports 5060 and 1000-2, on the box that asterisk is running, and on your NAT router. Dont port forward. Then in sip.conf [202] username=202 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=yes canreinvite=no [200] username=200 secret=X type=friend host=dynamic disallow=all allow=g729 allow=alaw allow=ulaw context=interne nat=yes canreinvite=no then restart linksys and thomson, and you will see that they both register on asterisk cli. Now you will be able to call/receive on both. Thanks for your answer, but if i don't put a port forward, i have : 200/20083.167.122.119 D N 5060 UNREACHABLE On the thomson, i have SIP Unregister, it's a important option ? Thanks bye ___ --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] Dropping extra frame of G.729 ?
Hi anyone know where i can solve this problems ? : Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Thanks ___ --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] Outgoing Call group ?
Hi
it's possible to create a group of outgoing dial ?
For exemple:
exten = _0.,1,Dial(SIP/voip1/${EXTEN:1},90,rt)
exten = _0.,2,Hangup
exten = _0.,1,Dial(SIP/voip2/${EXTEN:1},90,rt)
exten = _0.,2,Hangup
and when my user call, if voip1 are used, he use voip2
and use not the same line.
Thanks bye
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[asterisk-users] Problems compil 1.2.11
Hi when i want compile asterisk 1.2.11, i have this error : make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime' cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6 ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc -O6 ) works... no configure: error: installation or configuration problem: C compiler cannot create executables. make: *** [editline/libedit.a] Erreur 1 [EMAIL PROTECTED] asterisk-1.2.11]# what is the library that i don't have put on my server ? ___ --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] Problems compil 1.2.11
Anyone have a idea ? Noc Phibee a écrit : Hi when i want compile asterisk 1.2.11, i have this error : make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime' cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6 ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc -O6 ) works... no configure: error: installation or configuration problem: C compiler cannot create executables. make: *** [editline/libedit.a] Erreur 1 [EMAIL PROTECTED] asterisk-1.2.11]# what is the library that i don't have put on my server ? ___ --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] Help please == Wrong password
Hi i have a small problems with my asterisk connected to phonesystems : Now i have this message: -- SIP read from 62.39.136.151:5060: SIP/2.0 403 Cant accept register from myself Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060 From: sip:[EMAIL PROTECTED];tag=as42b95c05 To: sip:[EMAIL PROTECTED];tag=e3fe971527b049ab0c1e91db33fcbf5f.cf8c Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Server: PSN Sip Proxy (1.1.3 (PRX3-EXTERNAL)) Content-Length: 0 Warning: 392 62.39.136.151:5060 Noisy feedback tells: pid=11434 req_src_ip=62.39.136.151 req_src_port=5060 in_uri=sip:sip3.phonesystems.net out_uri=sip:sip3.phonesystems.net via_cnt==2 --- (9 headers 0 lines)--- Aug 30 17:12:50 WARNING[15568]: chan_sip.c:10010 handle_response: Forbidden - wrong password on authentication for REGISTER but my login/password are correct into sip.conf the configuration have changed in asterisk 1.2.11 ? thanks for your help ___ --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] Asterisk = Master and Slave ?
Hi a small question: I have one Asterisk Server with: VoIP Provider gateway for incomming/outgoing call 5 VoIP Phone (i name it Master) i want add a another Asterisk server but only connected to: 5 new VoIP Phone To the master for incoming/outgoing call (in g729) It's possible ? anyone have a sample of config ? thanks ___ --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] Asterisk and Phonesystems ...
Hi on a new Asterisk installation, i have a small problems with Asterisk and the VoIP Operator PhoneSystems. Anyone have connected Asterisk to Phonesystems ? I have this when i want call: chan_sip.c:9696 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'Jerome sip:[EMAIL PROTECTED];tag=as341491ez' jerome are the name that i have put on my SIP Phone connected to Asterisk ___ --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] increase the volume ?
Tzafrir Cohen a écrit : On Thu, Jun 08, 2006 at 02:12:48PM +0200, Noc Phibee wrote: Hi, Is it possible de tell asterisk to increase the volume? When we place or recieve a call the volume is very low, using a smartphone or a hardphone. What phone is it, exactly? Thomson Speedtouch 2030 (same that Alcatel) ___ --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] increase the volume ?
Martin Joseph a écrit : On Jun 8, 2006, at 10:26 PM, BJ Weschke wrote: On 6/9/06, Noc Phibee [EMAIL PROTECTED] wrote: anyone have a answer at this question ? Noc Phibee a écrit : Hi, Is it possible de tell asterisk to increase the volume? When we place or recieve a call the volume is very low, using a smartphone or a hardphone. Use the 'txgain' and 'rxgain' parameters in the CHANNEL dialplan function that's now in /trunk to turn up the volume. If you are talking about Zap channels. Marty I don't have ZAP, i use only a VoIP Provider without cards ;=) ___ --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] increase the volume ?
Hi, Is it possible de tell asterisk to increase the volume? When we place or recieve a call the volume is very low, using a smartphone or a hardphone. Thanks for advance ___ --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] increase the volume ?
anyone have a answer at this question ? Noc Phibee a écrit : Hi, Is it possible de tell asterisk to increase the volume? When we place or recieve a call the volume is very low, using a smartphone or a hardphone. Thanks for advance ___ --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] Asterisk and Fax ?
Hi it's possible that send and receive (receive in priority) a fax with Asterisk without card ? I am very interessed by a solution for receive the fax, convert in pdf and sent to email Thanks for your help ___ --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] Upgrade Cisco 7910 with Asterisk ?
Hi it's possible to upgrade the firmware of a cisco 7910 with asterisk ? he have a other solution for upgrade it without callmanager ? thansk for your help ___ --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] Trunk SIP howto ?
Hi anyone know if a Trunk SIP howto are created ? I have 8 VoIP account with for all 1 login/pass per number. i want add into my asterisk but not know where ;=) Other questions: my supplierhave a dns:sip.phonesystems.net this name have 2 IP address it's not a problems for Asterisk that he have registred on the first ip address and receive information of the second ? Thanks for your help ___ --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] Upgrade Cisco 7910 with Asterisk ?
Thanks sergio for your answer. But cisco france say me that i cant' bye SmartNet contract on this product. Only one solution are possible: Bye a special contract at $180.00 ... Pff i can bye a new equipment with this price hihihi i can't guest the latest firmware, for me i thinks that the solution are buy new voip phone and put the 7910 in Dead If anyone know a solution for get the latest firmware, mail me Bye Sergio Chersovani a écrit : Noc Phibee ha scritto: it's possible to upgrade the firmware of a cisco 7910 with asterisk ? You need the legal firmware upgrade file download the chan_sccp code from http://chan-sccp.berlios.de configure it and use the imageversion param to upgradde the phone firmware. Of course you need a tftpserver and if you run a tftpserver you just need a SEPmac to upgrade the phone So the correct answer is: you don't need a CCM nor asterisk to upgrade a cisco phone firmware. You just need the firmware file, a tftpserver and a configuration file (SEPmac) take a look here http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Sergio ___ --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] Upgrade Cisco 7910 with Asterisk ?
Thanks all for your answer ... all smartnet contrat have access to all firmware in voip ? thanks Ryan Amos a écrit : Cisco phones are not ideal for single-phone setups. If you were to have a lot of them, a $180 support contract is no big deal... However, for Europeans, there should be an $8 online-only support contract that gives you access to file downloads only. Being an American, This should be enough, however if you are only wanting a small number of phones you might want to look elsewhere. The main advantage of Cisco's phones comes when installing a large number of them, as the central management is ideal in an office PBX environment. Try this part number though: CON-SNT-PKG1-VS Supposedly costs 66 euros from wstore.fr (I found this in an old e-mail asking about smartnet contracts on the chan_sccp mailing lists.) Best of luck! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noc Phibee Sent: Monday, November 28, 2005 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ? Thanks sergio for your answer. But cisco france say me that i cant' bye SmartNet contract on this product. Only one solution are possible: Bye a special contract at $180.00 ... Pff i can bye a new equipment with this price hihihi i can't guest the latest firmware, for me i thinks that the solution are buy new voip phone and put the 7910 in Dead If anyone know a solution for get the latest firmware, mail me Bye ___ --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] SIP Trunk in incoming ? it's possible ?
Hi i renew my question ;=) i have 8 phone number provided by my VoIP supplier : 081037XX0 081037XX1 081037XX2 ... For each, i have a login/password where in put the registrer into my config ? it's a Trunk on incoming no ? i have put one register= per number but that's don't work. Thanks for your help ___ --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] Asterisk and Cisco Phone 7910
Hi i have buy a used Cisco Phone 7910 for use with my asterisk. The firmware version are 3.2(2.8), it's good for connect to asterisk ? For update the fiormware, where i can get a new firmware ? thanks bye ___ --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
