[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

Re: [asterisk-users] 2 Questions: Answer with music don't work and Voicemail direct access ?

/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

[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:

[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

[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:

[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

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

[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 ;=) ___

[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

[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:

[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 --

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

[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:

[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

[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 )

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

[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

[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

[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

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

[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

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

[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:

[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

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

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

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

[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

[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

[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

Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

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

Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

: [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

[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

[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

[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

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

[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

[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

[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

[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

[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

[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

Re: [asterisk-users] AdvancedVoIP Billing ?

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

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

[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

[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

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

Re: [asterisk-users] Asterisk and TDM400P ?

[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

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

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

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

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

[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:

[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] 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

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

[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

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

[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

[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

[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

[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 =