[Asterisk-Users] ACD with polycom ip phones

Hello Can we find a patch for asterisk-1.2.2 in order to test ACD with polycom phones ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs

RE: [Asterisk-Users] No audio? Update your Asterisk

hello, Could you give us the path to the patch quickly ? Harry Gaillac --- Olle E Johansson [EMAIL PROTECTED] a écrit : This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem

RE: [Asterisk-Users] No audio? Update your Asterisk

hello, Could provide us how to fix this serious bug my server is out of order please to post how to solve quickly this problem . Harry --- Olle E Johansson [EMAIL PROTECTED] a écrit : This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark

Re: [Asterisk-Users] No audio? Update your Asterisk

Hi all asterisk users On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Olle E Johansson wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please

Re: [Asterisk-Users] ACD with polycom ip phones

Hello, Can you provide a patch from your special branch for asterisk-1.2.3 ? can you post a how-to ? Even these features won't be include in th main branche a patch should be available. Regards harry --- BJ Weschke [EMAIL PROTECTED] a écrit : On 1/25/06, Douglas Garstang [EMAIL PROTECTED]

RE: [Asterisk-Users] ACD with polycom ip phones

Please to post a patch for asterisk-1.2.3 for polycom_acd_functions . Regards harry --- Douglas Garstang [EMAIL PROTECTED] a écrit : Oo I thought that this was a Polycom limitation, rather than an Asterisk one. Your saying that the Polycom phones would receive SIP signalling that

[Asterisk-Users] Monitoring

Hi asterisk and ser users, Is there a solution to monitor asterisk and ser with snmp ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs

[Asterisk-Users] file.c:509 ast_openstream_full: File 100 does not exist in any format

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]:

[Asterisk-Users] How many digium cards per server ?

Hello, How many digium cards is supported per asterisk server ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour

[Asterisk-Users] app_snmp

Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et

RE: [Asterisk-Users] cdrtool

Hello, Call sip:[EMAIL PROTECTED] Regards harry --- Jimmy Smith [EMAIL PROTECTED] a écrit : anyone having weird problems on latest cdrtool? #!/usr/bin/php4 *Fatal error*: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in

[Asterisk-Users] app_snmp

Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et

[Asterisk-Users] file.c:509 ast_openstream_full: File 100 does not exist in any format

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]:

Re: [Asterisk-Users] Polycom 501 ACDlogin

Can we patch the stable release with your SVN branch ? Regards Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have

[Asterisk-Users] Redirect a sip outbound requests to a sip proxy

Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar

[Asterisk-Users] Outbound Proxy Support

Hi all, May I have to patch asterisk-1.2.x with this patch http://bugs.digium.com/bug_view_page.php?bug_id=0002859 to configure an outbound sip proxy in sip.conf ? Regards Harry ___

[Asterisk-Users] Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:[EMAIL

Re: [Asterisk-Users] MWI, SER and asterisk

Hello, You can use ser as an outbound sip proxy and asterisk as a register server . Your sip agents will get MWI, ... Harry --- Christian B [EMAIL PROTECTED] a écrit : Hi Sharon! This is pretty difficult, i was not able to implement it so far(though my ser-skills are pretty basic). At

[Asterisk-Users] app_queue and ARA

Hello, I've configured ACD with ARA asterisk-1.2.4 . I try show queues command but no queue is shown. why ? Can I keep the caller on queue until an agent answer the call ? I use ARA to configure queues and members however i have to use agents.conf to store the agents. I wish to configure

[Asterisk-Users] TIMEOUT(s)

Hello, Here is part of my extensions.conf. I set both absolute and response timeouts according to the day context. I wish to asterisk hangup after 60s and 10s to play or replay the annoucement . Asterisk doesn't jump to T extension. How can fiox this problem ? harry ... [day] exten =

[Asterisk-Users] Setting up announcement on reply to 4xx 5xx 6xx messages

Hello, I wish to play a recorded announcement on reply to 4xx 5xx 6xx messages . According to the status a audio file would be played from asterisk server via ser to the caller How can I configure a such feature ? My configuration: Ser act as an outbound sip proxy . Asterisk a sip media

RE: [Asterisk-Users] Asterisk behind NAT

Hello, I advise you to install open(ser) with natelper module. Harry --- Kerry Garrison [EMAIL PROTECTED] a écrit : Yes. In Sip.conf you need the following lines: externip=xxx.xxx.xxx.xxx ; put public ip address here localnet=192.168.10.0/255.255.255.0 ; edit as appropriate In your

[Asterisk-Users] 407 proxy authentication

Hello, Asterisk sent back 407 proxy authentication . How can avoid this ? I set insecure=very without success in sip.conf and my sql server . Harry ___ Nouveau : téléphonez moins cher

[Asterisk-Users] HELP !!!!!

Hello, I wish to set a sip uri sip:[EMAIL PROTECTED] I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten = info,1,Answer() exten =

Re: [Asterisk-Users] HELP !!!!!

Jeremy I set in sip.conf [general] context=sip and [sip] include = info include = support [info] exten = info,1,Answer() exten = info,n,Dial(Sip/84,10) exten = info,n,Dial(Sip/85,10) exten = info,n,Hangup where info and support are hunt group --- Jeremy McNamara [EMAIL PROTECTED] a

Re: [Asterisk-Users] HELP !!!!!

Yes I reload and restart it --- Jeremy McNamara [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: I set in sip.conf And you have reloaded asterisk, right? Jeremy McNamara

[Asterisk-Users] HELP !!!!!

Hello, I wish to set a sip uri sip:[EMAIL PROTECTED] I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten = info,1,Answer() exten =

[Asterisk-Users] 407 proxy authentication

Hello, look at this I can't receive calls from other domains I wish sip:[EMAIL PROTECTED] are forwarded to asterisk however this one spend its time to ask 407 proxy authentication. asterisk 1.2.5 + realtime how can i fix this problem what' wrong ? extension.conf [info] exten =

[Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c

Tzafrir, How did you set sip:[EMAIL PROTECTED] I use serasterisk look at my sip.conf and extensions.conf Regards Harry [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay

[Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c

Tzafrir, How did you set sip:[EMAIL PROTECTED] I use serasterisk look at my sip.conf and extensions.conf Regards Harry [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay

[Asterisk-Users] Disable 407 proxy authentication for outbound domains

Hello, I posted a lot of mails may be asterisk is not able to accept sip calls from internet !? My english is not fluent i try my best ! My problem I use ser+asterisk. For local calls there are no problem (PSTN or IP) Now i wish to receive calls from other internet domain but asterisk ask

[Asterisk-Users] Call me for testing my system

Dear User, Anybody could dial these sip uri : sip:[EMAIL PROTECTED] (french) sip:[EMAIL PROTECTED] (music 60s) sip:[EMAIL PROTECTED] (french) Thanks for help ___ Nouveau : téléphonez

Re: [Asterisk-Users] Call me for testing my system

Thanks Thomas, I could not hear you too ! may be the firewall Harry --- Thomas Winter [EMAIL PROTECTED] a écrit : On Monday 10 April 2006 11:59, [EMAIL PROTECTED] wrote: Dear User, Anybody could dial these sip uri : sip:[EMAIL PROTECTED] (french) sip:[EMAIL PROTECTED] (music

Re: [Asterisk-Users] Call me for testing my system

Could you try again please? --- Thomas Winter [EMAIL PROTECTED] a écrit : On Monday 10 April 2006 11:59, [EMAIL PROTECTED] wrote: Dear User, Anybody could dial these sip uri : sip:[EMAIL PROTECTED] (french) sip:[EMAIL PROTECTED] (music 60s) sip:[EMAIL PROTECTED] (french)

[Asterisk-Users] Polycom Microbrowser

Hello, I read the polycom microbrowser post here http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser Can we access a webmail application like horde/imp or others (which ones) to read and listen voicemails , send e-mails, ... ? Regards Harry

RE: [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message

Hello, I use ser for IM and presence and asterisk When my sip agents send REGISTER messages I have two records one in ser database the other in asterisk database . Ser manage far-end nat IM and presence (SIMPLE). Harry --- Douglas Garstang [EMAIL PROTECTED] a écrit : I don't think Asterisk

[Asterisk-Users] asterisk + mobicents

Hello, I look at the mobicents project. Somebody has experience within both projects ? Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement

[Asterisk-Users] Setting up a t38 fax gateway

Hello to all, Is there an how-to for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server.

[Asterisk-Users] SIPredirect

Hello, I read http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect I wish to configure asterisk as a redirect server. I have badly understood this command . ASTERISK | sip agents nated ==SER When sip agents send INVITE to the

RE: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Hi all, Where can we find a roadmap of asterisk 1.4 release ? Harry --- Olle E Johansson [EMAIL PROTECTED] a écrit : Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical

[Asterisk-Users] compiling zaptel-1.2.5

Hello, What's wrong ? make install . options torisa base=0xd alias char-major-196 torisa alias wcfxs wctdm alias wct2xxp wct4xxp if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: ***

RE: [Asterisk-Users] compiling zaptel-1.2.5 [SOLVED]

--- [EMAIL PROTECTED] a écrit : Hello, What's wrong ? make install . options torisa base=0xd alias char-major-196 torisa alias wcfxs wctdm alias wct2xxp wct4xxp if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in

RE: [Asterisk-Users] SIPredirect [2]

--- [EMAIL PROTECTED] a écrit : Hello, I read http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect I wish to configure asterisk as a redirect server. I have badly understood this command . ASTERISK | sip agents nated

RE: [Asterisk-Users] Setting up a t38 fax gateway [2]

--- [EMAIL PROTECTED] a écrit : Hello to all, Is there an how-to for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards

Re: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Ok, Im not a developper but what do you think of both a wish list . Harry To answer your question: there is no roadmap for 1.4. We just began the 'scheduled release' cycle with this release, and we are still trying to feel our way into the process and learn how much work we can

[Asterisk-Users] Sip t38 gateway tests

Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver

[Asterisk-Users] Re: [Serusers] Sip t38 gateway tests

Thanks for these informations I would have prefer to receive them from asterisk-users instead of serusers !! May be they are sleeping . Ok i have not installed spandsp because of i don't find some scripts like in hylafax for mail2fax fax2mail i've just patched chan_sip.c Regards Harry ---

[Asterisk-Users] URGENTS: seek people for video tests with asterisk/ser/rtpproxy + eyebeam

Hi asterisk, openser, ser users. I have to check video support between asterisk, open(ser) and rtpproxy . ASTERISK (b2bua+registrar server) | | | | SER + rtpproxy | | NAT | | sip agents (with video support)

[Asterisk-Users] chan_sip.c patched with t.38

Hello, Is there Somebody to provide me a DID numder on a voip gateway which one support t.38 to test FOIP ? Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour

RE: [Asterisk-Users] Early media after a dial command

Hi Benjamin, How do you setup early media in asterisk ? Harry --- Benjamin Lawetz [EMAIL PROTECTED] a écrit : Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the

[Asterisk-Users] Which h323 channel for asterisk and gnugk ?

Hello, I need to install a h323 channel in order to asterisk act as a sip/h323 translator . I want to use gnugk in full proxy mode for the h323 terminals nated . Which h323 channel for asterisk and gnugk h323 oh323 or ooh323c ? Harry

[Asterisk-Users] asterisk monitoring / res_snmp

Hello, I 've installed both cacti and res_snmp for monitoring. Does res_snmp is able to send snmp traps when hardware is out of service or others status ? Harry ___ Faites de Yahoo!

RE: [Asterisk-Users] asterisk monitoring / res_snmp [2]

Does digium provide a snmp solution to monitor their telephony cards ? Harry --- [EMAIL PROTECTED] a écrit : Hello, I 've installed both cacti and res_snmp for monitoring. Does res_snmp is able to send snmp traps when hardware is out of service or others status ? Harry

Re: [Asterisk-Users] asterisk monitoring / res_snmp [2]

Is it a solution to add some code in those cards in order to a snmp agent could get/query some informations about the state of the cards ? Do you know cards with snmp support ? Harry --- Kevin P. Fleming [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: Does digium provide a snmp solution

[Asterisk-Users] monitoring sangoma cards via snmp

Hello, Digium does not provide snmp support to monitor their cards ! Anybody has tried Sangoma product A104 Quad T1/E1 or others ? Regards harry ___ Yahoo! Mail réinvente le mail !

[Asterisk-Users] call parked / MOH

Hello, How can I park a call or put on hold a caller from an analogue to sip agents ? PSTN===FXO/asterisk=sip agents When I press hold key or #800 the channel is hangup ?? Harry Regards

URGENT please [Asterisk-Users] call parked / MOH

Hello, How can I park a call or put on hold a caller from an analogue to sip agents ? PSTN===FXO/asterisk=sip agents When I press hold key or #800 the channel is hangup ?? Harry Regards

URGENT please [Asterisk-Users] call parked / MOH

Hello, How can I park a call or put on hold a caller from an analogue to sip agents ? PSTN===FXO/asterisk=sip agents When I press hold key or #800 the channel is hangup ?? Harry Regards

URGENT please [Asterisk-Users] call parked / MOH

Hello, How can I park a call or put on hold a caller from an analogue to sip agents ? PSTN===FXO/asterisk=sip agents When I press hold key or #800 the channel is hangup ?? Harry Regards

[Asterisk-Users] RE: snmp and asterisk

hi david, can you explain me this please? If Sangoma hardware support snmp ithink it would be a better choice than digium . How can we know the state of the sangoma cards with an snmp agent ? Harry --- David Yat Sin [EMAIL PROTECTED] a écrit : Hi Harry, The Sangoma Card when used for TDM

[Asterisk-Users] RE: snmp and asterisk

I use res_snmp.so with asterisk do you provide mib --- David Yat Sin [EMAIL PROTECTED] a écrit : Hi Harry, The Sangoma Card when used for TDM Voice will work under zaptel, so you would need to perform the SNMP through Asterisk. Regards, David Yat Sin Sangoma Technologies (905) 474 1990

[Asterisk-Users] parking a call /put on hold

Hello, Can we park a call or put on hold a caller in a queue ? I have sip polycom phone but when i press hold key or #800 i can't neither park call nor hold this call . Is it possible ? Harry

[Asterisk-Users] Re: [asterisk-dev] SNMP support for Digium Cards

Can you explain me why I 'm obnoxious cretin ? I 've been asking for monitoring the digium cards via snmp. What's the problem ? I post to asterisk-users and asterisk-dev to get informations why some people of these list insult me ? What are yours problems ? Plesae to send me back my cretin

[Asterisk-Users] Re: [asterisk-dev] SNMP support for Digium Cards

Can you explain me why I 'm obnoxious cretin ? I 've been asking for monitoring the digium cards via snmp. What's the problem ? I post to asterisk-users and asterisk-dev to get informations why some people of these list insult me ? What are yours problems ? Plesae to send me back my cretin

[Asterisk-Users] Re: [asterisk-dev] SNMP support for Digium Cards

Tzafrir, cross-posting to asterisk-users and to asterisk-dev is not a good idea. You should know that by now, as you have been told that numerous time. I cross-posting because nobody answer ! I think some people are able to answer this question ? Please, don't tell me people here are not

[Asterisk-Users] Re: [asterisk-dev] SNMP support for Digium Cards

Tzafrir, cross-posting to asterisk-users and to asterisk-dev is not a good idea. You should know that by now, as you have been told that numerous time. I cross-posting because nobody answer ! I think some people are able to answer this question ? Please, don't tell me people here are not

[Asterisk-Users] snmp for asterisk

Hi to all, I've ever post many times some questions about snmp to monitor asterisk . I need to be adviced to extend res_snmp in order to monitor both hardware and softs of asterisk . I wish to monitor digium cards to get call and line statistics as well as status and errors (traps). Which

Re: [Asterisk-Users] snmp for asterisk

Are you interesting in monitoring asterisk with snmp before i translate the text in english ? Harry --- Michael Labuschke [EMAIL PROTECTED] a écrit : Rich Adamson schrieb: Harry, I've ever post many times some questions about snmp to monitor asterisk . I need to be adviced to

[Asterisk-Users] Please help.. I need a h323 user for tests

hello, Is there somebody wit a h323 terminal ? ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails,

[Asterisk-Users] need help

hello, I have to test asterisk/gnugk is their somebody, sur cette putain de liste, with a h323 terminal ? harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver

Re: [Asterisk-Users] need help

is their nobody here with a h323 terminal, netmmeting ... I just need a h323 terminal register with asterisk/oh323/gnugk just five minutes just aggressive because of I'm feeling tired --- Administrator TOOTAI [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: hello, I have to

[Asterisk-Users] Tr: Re: The OpenNMS Group, Inc.: opennms and asterisk pbx

Remarque : message transféré en pièce jointe. Hello , To people who told me I'm cretin !! Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos

RE: [Asterisk-Users] Asterisk as a proxy

NO ! --- ram [EMAIL PROTECTED] a écrit : Hi does asterisk act as SIP proxy ?, like SER any documents if does, will be great help ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

RE : [Asterisk-Users] Polycom-Asterisk hints/presence

Hello, Try both asterisk and ser for IM/presence . --- Damon Estep [EMAIL PROTECTED] a écrit : I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are watching other extensions would be notified when the

[Asterisk-Users] asterisk-1.2.9 is not stable

I upgrade 1.2.7-1 to 1.2.9 but asterisk is not stable I 've lost call SIP-ZAP. channels. i can't hear sound because of res_snmp.so . Is it a bêta release ?? I downgrade to 1.2.8 or 1.2.7 I do hope 1.4 will be a real stable realease Harry __ Do

RE : Re: [Asterisk-Users] asterisk-1.2.9 / res_snmp.so

--- [EMAIL PROTECTED] a écrit : hello, How asterisk could support res_snmp even this module don't help to monitor all asterisk features? monitoring asterisk with snmp would be a good thing. Which solution ? Harry --- Kristian Kielhofner [EMAIL PROTECTED] a écrit : [EMAIL

[Asterisk-Users] polycom ftp

Anydody need some access to polycom ftp server ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail

[Asterisk-Users] Polycom Files

Hello, If somebody need the latest Polycom Files contact me or look at ftp://nxs.yi.org Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités

[Asterisk-Users] dial pattern

Hello, I have to dial prefix 9 for non local numbers however when i missed calls i Can't redial this number because of 9 is not append . I use polycom phones . What Can i do ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous

[Asterisk-Users] SRTP/SIPS

Hello, Is there a project for SRTP/SIPS in Asterisk ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail

[Asterisk-Users] IP/SS7 gateway on Sun Ultra 20 amd64

Hello, I have to setup a IP/SS7 gateway on a Sun Ultra 20 Debian Sarge for AMD64 Can we compile asterisk on AMD64 processor ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les

RE : Re: [Asterisk-Users] asterisk on AMD 64 BIT

Hello John , What about debian sarge ? Harry --- John Millican [EMAIL PROTECTED] a écrit : I have 2 servers currently running 64 bit SuSE 10.x on AMD Opteron processors both of which are working very well. John Millican Senior Partner Director of Technology Sentinel Communications PO

[Asterisk-Users] Which simple billing application

Hello, I look at voip-info for a simple billing application . I wish to calculate price to pay according to the datas stored in cdr table (unixodbc/mysql). what do you advise me ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous

[Asterisk-Users] asterisk+cdrtool

hello, i try to install the latest cdrtool but i get this messsage: #!/usr/bin/php4 Fatal error: Cannot redeclare class db_sql in /var/www/CDRTool/phplib/db_mysql.inc on line 12 Thanks for help Harry __ Do You Yahoo!? En finir avec le spam?

[Asterisk-Users] subscription

___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

[Asterisk-Users] fax and voice

Hello, I wish to configure Hylafax in order to send either fax or voice to Asterisk I've got a TDM400P (1FXS/1FXO) . What' s the best way to check the line to send fax or voice for incoming or outgoing ? Thanks for help H.G

[Asterisk-Users] Fax detected, but no fax extension

Hello, I get this message when i send fax Fax detected, but no fax extension. I read mailing list . Can we solve this ? my conf : =PSTN==(fxo)-Asterisk-(fxs)==Hylafax==SMTP/POP3 server zapata.conf: context=fax faxdetect=both signalling=fxo_ks group=2 channel = 2 extension.conf [fax] exten =

RE: [Asterisk-Users] Fax detected, but no fax extension

OK, Is Asterisk able to switch incoming calls according to fax or voice to the right extension . Which function detect incoming signal ? Regards H.G --- Colin Anderson [EMAIL PROTECTED] a écrit : You need an extension called fax in your [fax] context like this: [fax] exten =

[Asterisk-Users] ChanIsAvail()

Hello, I configure a asterisk server with tdm400p . I wish to set chanisavail() in order to allow users or the hylafax server to dial numbers to pstn . however I can't write the rules to forward requests to the dial pattern when channel is available. I try this however priority 2 fail. how

Re: [Asterisk-Users] Fax detected, but no fax extension

--- JP Carballo [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: OK, Is Asterisk able to switch incoming calls according to fax or voice to the right extension . Which function detect incoming signal ? If you have faxdetect enabled in zapata.conf, (the default is off),

[Asterisk-Users] placing a call in one or several call groups

Hello, I read http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups So i set callgroup and pickupgroup in sip.conf . How can I forward an incoming call to one or more callgroup. Regards Harry

[Asterisk-Users] Can't pickup call when dialing *8 extension

Hello, I added callgroup=1 and pickupgroup=1 for sip channels however I can't pickup a call (see below ) between sip phones when i dial *8 . May I have to add app_pickup to solve this problem. I use asterisk-1.2 Regards Harry serveur1*CLI -- SIP read from 80.119.8.167:5060: ACK sip:[EMAIL

Re: [Asterisk-Users] Can't pickup call when dialing *8 extension

*8 is coded in res_features.so . What are the right extension to dial for pickup calls between sip=sip or zap=sip ... Harry --- Rich Adamson [EMAIL PROTECTED] a écrit : You might have to use *8#. At least I do with my 7960. I added callgroup=1 and

[Asterisk-Users] Can't pickup call when dialing *8 extension (resent)

*8 is coded in res_features.so . What are the right extension to dial for pickup calls between sip=sip or zap=sip ... Harry --- Rich Adamson radamson at routers.com a écrit : You might have to use *8#. At least I do with my 7960. I added callgroup=1 and

[Asterisk-Users] ACD with polycom ip phones

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger !

[Asterisk-Users] ACD with polycom ip phones (resent)

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger !

Re: [Asterisk-Users] ACD with polycom ip phones

When ACD is used the queues and agents are configured so agents have to send agent id and password to become available in a queue . Harry --- Matthew matthew@zeut.net a écrit : [EMAIL PROTECTED] wrote: Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom

Re: [Asterisk-Users] ACD with polycom ip phones

So we have to add a context like this to login/logout agents. I add 4 agent in a queue with roundrobin strategy . What's going on if the first available agent don't answer the call ? Asterisk-1.2 [agents] ;Agent Login exten= 501,1,AgentCallbackLogin(||[EMAIL PROTECTED]) ;Agent Logout exten=

[Asterisk-Users] NVFaxDetect

Hello, I have a single line to receive fax and voice. I add faxdetext in zapata.conf and [pstn] exten = s,1,Answer exten = s,2,Queue(MyQueue|tn||100) exten = s,3,Hangup exten = fax,1,Dial(Zap/g2) However when fax tone is detected both phones in queue and the modem of Hylafax server answer the

Re: [Asterisk-Users] ACD with polycom ip phones

Which standard for ACD login/logout ? --- Kevin P. Fleming [EMAIL PROTECTED] a écrit : Adam Goryachev wrote: Could chan_sip simply start executing the DP at a particular extension ?? or would that require the existence of a channel, which there isn't really since it is just XML not

  1   2   >