[asterisk-users] Sox and bad quality when converting to 8 kHz

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help best regards Thomas --

[asterisk-users] multiple devices wants to call through single peer (trunking)

Hi list, how can I set up an peer, so that behind one IP (NAT) multiple devices can access to this single peer to make outbound calls. Some of these multiple devices will be SIP phones and these SIP phones are trying to make registrations to this peer. best regards Thomas --

[asterisk-users] How to avoid AGI script is canceled if caller HangUp

Hi, is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. Some thing can happend with an Macro if caller hang up

[asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime

Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas --

[asterisk-users] change L(x[:y][:z]) parameter of DIAL command after call is bridged

Hi, is there any way from outside change x,y an z after a call is bridged? maybe with AMI interface? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] DIAL IAX2 vs. SIP

Hi, documentation shows me: Dial(Tech/User:passw...@host/Extension,Timeout,Optionen) This is working for IAX2. If Iam using DIAL(SIP/u...@secret@sip.domian.tls/123456) Asterisk shoes no host with name sip.domian.tls/123456 How to put in extension if using the DIAL command with userid and

[asterisk-users] Realtime difference sipusers sippeers

Hi, I would have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas

[asterisk-users] GROUP() decrement

Hi, how I can decrement the value of GROUP_COUNT() by one after I have before used GROUP(), so that other channel will get the correct value of GROUP_COUNT(). for examaple exten = _X!,n,Set(GROUP()=${Provider}) exten = _X!,n,DIAL(SIP/${ext...@${provider}) When Dialstatus is CONGESTION I want

[asterisk-users] line disconnected after 20 seconds no reply to our critical packet

Dear all, I have from time to time problems with disconnect after exact 20 seconds. I have these problems from time to time in LAN after using PickUP() with 1.2 I have these problems from time to time in WAN when the internet connection is not reliable with 1.4 Is there any way to fix it?

Re: [asterisk-users] Codec problems when using G.723

On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote: The best (and maybe only way) is to set your client and your service provider to only do G.723. Really, thats not the way it should work. How I can find out the codec of an incomming call? Is there any way to use ${SIP_CODEC} to

Re: [asterisk-users] Codec problems when using G.723

On Monday 10 November 2008 16:52, Eric ManxPower Wieling wrote: Thomas Winter wrote: On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote: The best (and maybe only way) is to set your client and your service provider to only do G.723. Really, thats not the way it should work

[asterisk-users] Codec problems when using G.723

Hi, I have a problem with codecs. I have an provider with allowed codec alaw, ulaw, g.723 I have SIP clients with codec allowed alaw, ulaw, g.723 If a SIP clients wants call through with g.723 Asterisk is using alaw to connect to the provider, so its not working because only passthrough would

Re: [asterisk-users] realtime queue reload

On Tuesday 09 September 2008 12:30, Atis Lezdins wrote: On Tue, Sep 9, 2008 at 1:22 PM, Thomas Winter [EMAIL PROTECTED] wrote: On Monday 08 September 2008 14:44, Atis Lezdins wrote: On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED] wrote: I dont have problem to make

Re: [asterisk-users] realtime queue reload

On Monday 08 September 2008 14:44, Atis Lezdins wrote: On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED] wrote: I dont have problem to make a reload by AMI. My questions was if module reload app_queue.so is the right way to do this, because whis reload I reload everything

Re: [asterisk-users] realtime queue reload

On Saturday 06 September 2008 21:47, Brian wrote: Hi Thomas, The queue definitions and its member list will be reloaded each time a caller joins the queue. So you don't need to reload it manually. Hi, is not work for periodic-announce-frequency and periodic-announce. An reload is

Re: [asterisk-users] realtime queue reload

On Sunday 07 September 2008 21:49, Atis Lezdins wrote: On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED] wrote: is not work for periodic-announce-frequency and periodic-announce. An reload is necessary. Asterisk is 1.4.21.1 It shouldn't be necessary. However you can try

[asterisk-users] realtime queue reload

Hi, Iam using queues through realtime, works fine. After making changes how do I make Asterisk aware? Is module reload app_queue.so through AMI the correct way to do this? best regards Thomas ___ -- Bandwidth and Colocation Provided by

[asterisk-users] vars in Macros called by DIAL with option M

Hi all, Iam using an DIAL Command wird Macro if callee is answer the call. exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect)) exten = 123,n,NoOp( ${var_from_macro}) In Macro test_connect Iam generating an new variable var_from_macro and would like to use this var in the original

Re: [asterisk-users] vars in Macros called by DIAL with option M

On Tuesday 05 August 2008 18:04, Tilghman Lesher wrote: On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote: Hi all, Iam using an DIAL Command wird Macro if callee is answer the call. exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect)) exten = 123,n,NoOp

Re: [asterisk-users] vars in Macros called by DIAL with option M

On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote: I don't think you can do that because, asterisk, in the caller thread will only read MACRO_RESULT to know if he has to connect the call or not. A workaround will be to : 1. before the dial, add a row in a database table and retrieve an id

Re: [asterisk-users] vars in Macros called by DIAL with option M

Lesher wrote: On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote: Hi all, Iam using an DIAL Command wird Macro if callee is answer the call. exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect)) exten = 123,n,NoOp( ${var_from_macro}) In Macro test_connect Iam

Re: [asterisk-users] Queue with different music for each caller

) then it will work in older Asterisk versions! br, Martin - Original Message - From: Thomas Winter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 24, 2008 5:50 PM Subject: Re: [asterisk-users] Queue

[asterisk-users] queues and MEMBERINTERFACE for AGI script

Hi, iam using and queue and starting an AGI script after caller connected to agent. How to find out in the script the connected agent, MEMBERINTERFACE seemed to be not work, either as variable in the queue command and also not in the AGI script. How to found out which agent is connected to

[asterisk-users] Queue with different music for each caller

Hi, is there an possibilty to have for each caller different music when queued. I see there only the global musiconhold = default in queues.conf, what menas same musci for all waiting callers. Any other idea to realize this? best regards Thomas ___

Re: [asterisk-users] Queue with different music for each caller

On Tuesday 24 June 2008 15:22, Martin Schrott - thinking:systems wrote: Hello Thomas you can use different music for each caller if you like. in extensions.conf you can set the music class. exten = s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller) Hi Martin, thanks for your suggestion,

[asterisk-users] custom functions is voicemail

Hi, I want to add some custom functions in voicemail. For example user can switch SMS on/off or the voicemail global on/off. Whats best way to do this? modify app_voicemail.c or or do everything in dialplan? or any other solutions (Asterisk 1.2.X please) best regards Thomas

Re: [asterisk-users] queue delay between calls to agents

On Thursday 05 June 2008 01:09, Tariq .. wrote: you can reduce the 5 seconds to any number you wish.. but from a personal experience .. if you put the retry to zero.. nothing will change.. i suggest to use 1 as your minimum aiting number Tarek Sawah thanks, retry = 1 is working retry = 0 looks

Re: [asterisk-users] Queue is sending calls to Agents even when they are in use

On Tuesday 03 June 2008 23:22, Atis Lezdins wrote: chan_agent with AgentCallbackLogin was working but not completely stable for my dialplan which was quite heavy when I was on 1.2, however you may try that out. Or just upgrade to 1.4 (or even 1.6 and try state_interface) Iam using API Action

[asterisk-users] queue delay between calls to agents

Hi, I want to reduce the dead time before the queue is calling the next agent. I see there 5 seconds delay. It is possible to reduce this time, or what is Asterisk doing within this timeframe. best regards Thomas ___ -- Bandwidth and Colocation

[asterisk-users] applicationmap in features.conf Asterisk 1.2 is ignoring DIAL tT options

Hi, I found out that GoTo in applicationmap is not working. OK, LOCAL is working. but I expected that applicationmap is using the DIAL option tT. But it doesnt, it works without tT Option, so also callee can use internal functions if callee knows the code. Any workaround avaiable? best

[asterisk-users] overlap calls from NT-BRI timeout problem

Hi, Iam getting calls from an POTS system on an NT port. Multiport BRI card running bristuff 0.3. From time to time the recognized number is incomplete and dial failed. Is there any way to increase timeout waiting for called numbers? Because dialed numbers can be from 3 to 13 digits there is no

Re: [asterisk-users] mpg123 on Thecus N2100

On Sunday 11 November 2007 01:38, Tzafrir Cohen wrote: On Sat, Nov 10, 2007 at 01:40:23PM -0600, Eric ManxPower Wieling wrote: Thomas Winter wrote: Hi, Iam running debian etch on thecus n2100 (Xscale 80219) I do not have MoH because standard mpg123 gives only loud noise. I can

[asterisk-users] mpg123 on Thecus N2100

Hi, Iam running debian etch on thecus n2100 (Xscale 80219) I do not have MoH because standard mpg123 gives only loud noise. I can not compile mpg123 from asterisk because of option -m486. Any way to get MoH running on this board. ___ --Bandwidth and

Re: [asterisk-users] mpg123 on Thecus N2100

On Saturday 10 November 2007 20:40, Eric ManxPower Wieling wrote: Thomas Winter wrote: Hi, Iam running debian etch on thecus n2100 (Xscale 80219) I do not have MoH because standard mpg123 gives only loud noise. I can not compile mpg123 from asterisk because of option -m486. Any way

[asterisk-users] bristuff: music on hold but no dialoptions tT defined.

Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22

Re: [asterisk-users] German SIP and/or IAX providers?

On Friday 12 October 2007 04:38, Ken D'Ambrosio wrote: Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. However, I'd need a few things: - Reliability. Can't have my branch office's DID's just going down. A then you

Re: [asterisk-users] bristuff for hfc card on Xscale 80219

On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote: On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote: Hi, Frozen or crashed? Do you see the console of the system? serial console is dead. kernel is 2.6.18-4 debian Etch. bristuff is latest zaptel-1.2.19 and asterisk-1.2.22 I

[asterisk-users] bristuff for hfc card on Xscale 80219

Hi, compile and load of modules works fine. After ztcfg I can see . . Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to HDLC with FCS check and then the board is frozen.

[asterisk-users] hints howto reset if wrong subscription Asterisk 1.2

Hi, sometimes Asterisk told me in the subscription: status confirmed so LED is on if the softphone is disconnected or the registration has expired. So the whole weekend LEDs have the wrong status. Manager Command Extensionstate is working correct, only the subscription is wrong. How can I

Re: [asterisk-users] How can I improve call quality?

On Friday 20 April 2007 20:01, Adrian Marsh wrote: Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP

Re: [asterisk-users] Play audio and continue to next priority before audio ends...

Am Monday 09 April 2007 23:20 schrieb Alejandro Mejía: Hello list members. I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because

[asterisk-users] Manager Originate and Var to long

Hi, I use Originate to make a call. I have problems to bring my vars into the channel. Are there restrictions more then only 24 vars at mentioned at www.voip-info.org? Any workaround to get this running? WARNING[4641]: manager.c:1365 get_input: Dumping long line with no return from

[asterisk-users] res_musiconhold.c:1243 load_module: No music on hold classes configured

Hi, I am using relatime for musiconhold.conf. After starting Asterisk I have to do an reload, otherwise no MoH is avaiable. Bug or do I have to change loading of modules in modules.con? ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] SIP provider did not send BYE if callee is an queue

Hi, I have an strange problem and did not understand what happened here. Iam using an SIP provider to call an analog POTS phone. I orginate the call to the analog phone and send the call in an queue. If the analog phone hangs up, the SIP provider did not send BYE and Asterisk think the line

Re: [asterisk-users] play file and action only stop if one defined key has been pressed

Am Friday 09 March 2007 23:51 schrieb Steve Murphy: On Fri, 2007-03-09 at 23:01 +0100, Thomas Winter wrote: I didnt see the option. The number can be different and is stored in mySQL exten = ${tmp_var},1,NoOp(INFO key pressed) exten = ${tmp_var},n,GoTo(s,restart) Woa! can you

[asterisk-users] play file and action only stop if one defined key has been pressed

Hi, I would like that user cann press 3 and then actions can be taken. Problem ist if the pressed key not 3 the user jumps to extension i and then the file will be played from start again. I would like that the play of file is only stopped if the user has pressed the key 3. What for an

Re: [asterisk-users] play file and action only stop if one defined key has been pressed

Am Friday 09 March 2007 22:27 schrieb Time Bandit: I would like that user cann press 3 and then actions can be taken. Problem ist if the pressed key not 3 the user jumps to extension i and then the file will be played from start again. I would like that the play of file is only stopped

[asterisk-users] queue information in mySQL

Hi, is it possible to have the information stored in /var/log/asterisk/queue_log realtime in mySQL? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] musiconhold.conf in realtime

Hi, I have problems with 1.2.14 and musiconhold.conf and realtime. I have to do moh reload at CLI to use the classes stored in mysql. Otherwise nothing is found if using SetMusicOnHold(${v_moh}) or DIAL with MoH. Are there known problems or how to get this running? thanks Thomas

[asterisk-users] Fwd: musiconhold.conf in realtime

If i do an asterisk -rx moh reload MoH stops and restarts on exsisting channels. If I do an moh reload through the Manager Interface Sound is dead on exsisting channels. Any other idea for an workaround? Hi, I have problems with 1.2.14 and musiconhold.conf and realtime. I have to do moh

Re: [asterisk-users] queues and LOCAL for members

Am Friday 02 February 2007 23:48 schrieb BJ Weschke: On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote: Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics

[asterisk-users] queues and LOCAL for members

Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. best regards Thomas

[asterisk-users] licence quick question

Hi, If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it as an complete product to an coustomer, do I have to put my developed code or the complete product under the GPL? best regards Thomas ___ --Bandwidth and Colocation

[asterisk-users] HFC-card and TDM400 with bristuff

Hi, is it possible to run an HFC-card with bristuff and an TDM400 in one PC? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Re: mySQL and to many connections with SQL statement UPDATE

Am Sunday 24 December 2006 16:53 schrieb JR Richardson: If Iam doing UPDATE SQL statements I got an overload for connection. am doing everytime an Disconnect ${connid}) but this is ignored. any idea? You must clear the resut ID and also issue a disconnect to the connection ID, see

[asterisk-users] mySQL and to many connections with SQL statement UPDATE

Hi, If Iam doing UPDATE SQL statements I got an overload for connection. am doing everytime an Disconnect ${connid}) but this is ignored. any idea? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp

Am Sunday 29 October 2006 01:31 schrieb Dovid B: Half asleep. Sorry for my last post. I believe you still need port forwarding for IAX. Time to keep to my bed time. If works as long as you have notransfer=no at both ends. Iam concerned that with SIP Asterisk is bridging up and I do not receive

[asterisk-users] Asterisk behind NAT and without portforwarding for rtp

Hi, I have an Asterisk behind NAT. NAT=yes and canreinvite=no in globals and for the peer. I call an peer. The peer advice to use another IP for the audio and my Asterisk is sending audio stream to the Audio server. Because of missing port forwarding I will not receive the audio stream and hear

[asterisk-users] Asterisk and ISDN and Hylafax

Hi, I have to set up an Asterisk with an 4-port BRI card. Hylafax should send and receive fax. Will this work reliable? Any recommandations for an 4-port BRI card? Other alternatives except analog fax units? thanks for your help best regards Thomas

Re: [asterisk-users] Asterisk and ISDN and Hylafax

Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes

Re: [asterisk-users] ChanIsAvail() in 1.2.12.1

Am Friday 06 October 2006 23:03 schrieb Douglas Garstang: *CLI -- Executing NoOp(SIP/3254101-0817a220, *** Originated call Chocolate Chip 3254101 - 3254103) in new stack -- Executing NoOp(SIP/3254101-0817a220, FOO1) in new stack -- Executing ChanIsAvail(SIP/3254101-0817a220, SIP/3254103)

[asterisk-users] Server for Asterisk PCI

Hi, can anybody recommend HP Proliant ML110 for Asterisk and ISDN interface cards? This Server has only two PCI 32Bit/33 MHz 3,3 Volt. Is this OK for PRI cards? thanks.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[asterisk-users] SIP user deny and permit for calls through Asterisk

Hi, i tried deny and permit in the peer definition. It works fine for registration purpose. But if the peer is dialing through Asterisk these settings are ignored. Only username and password are used for authentification. Is there anythink additional what I can use to prevent that the phone

[asterisk-users] sipbuddies realtime fields and latest documentation

Hi, I used voip-info.org for setup my realtime users. The mySQL table did not include for example the option call-limit. Where I can find information whats the correct field name to adjust my mySQL table? thanks Thomas ___ --Bandwidth and

Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)

Am Tuesday 25 April 2006 11:24 schrieb Olle E Johansson: 25 apr 2006 kl. 00.24 skrev Thomas Winter: Am Monday 24 April 2006 18:39 schrieb Doug Lytle: Thomas Winter wrote: Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP

[Asterisk-Users] SIP HEADER FROM: without CALLERID(name)

Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. I want to have the SIP HEADER like this: FROM: sip:CALLERID(number)@domain.tld thanks best regards Thomas

Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent

Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta: How do I report a Bug to Digium? or asterisk project? Did you report this bug? I checked and have seen only an timeout in the channel will kill the dead channels. Iam using GROUP_COUNT, so it is easy to kill my Asterisk if somebody is make

Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)

Am Monday 24 April 2006 18:39 schrieb Doug Lytle: Thomas Winter wrote: Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. Just a guess, try: SET

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Am Thursday 20 April 2006 01:21 schrieb tom: Thomas Winter wrote: I have done additional tests, because the documentation sample was not 100 % identical to my register command. OK: register = 44198:[EMAIL PROTECTED]/200 This jumps to 200, s is also working NOT OK: user:[EMAIL

[Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Hi, the documentation of sip.conf is telling me this: ;register = 1234:[EMAIL PROTECTED] ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

suggest if you want to bypass that problem, add /s (or whatever extension) to the register statement so you know for absolute sure that incoming calls on the registration will go to the extension that you expect. Aaron On Wed, 19 Apr 2006, Thomas Winter wrote: Hi, the documentation

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

register with one number and have it drop in on a totally different number in the context. register = 44198:[EMAIL PROTECTED]/200 Aaron On Wed, 19 Apr 2006, Thomas Winter wrote: Hi, [general] context=Sip_in register = 1234:[EMAIL PROTECTED]/s s is the same, it still looks

[Asterisk-Users] g.726 codec not working in one direction

Hi, Iam using Asterisk Asterisk 1.2.5 Iam calling: NOT OK: phone A -ulaw - Asterik-A - gsm - Asterisk-B - g.726 - POTS phone B NO sound from from phone A to phone B, phone B to phone A works If iam using ulaw to connect from Asterisk-B to POTS phone B everythink is OK: OK: phone A -ulaw -

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

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) Hi, sip:[EMAIL PROTECTED] (french) sip:[EMAIL PROTECTED] (music 60s) No Sound or voice!

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

On Monday 10 April 2006 13:49, [EMAIL PROTECTED] wrote: 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

Re: [Asterisk-Users] Group funcations not functioning

On Monday 10 April 2006 22:04, Raymond Chen wrote: Dear all, we have try to limit the outgoing channel by using GROUP() and GROUP_COUNT() to limit number of calls to a channel/trunk. but lately we upgraded to 1.2.5, 1.2.6 or SVN 1.2 , both functions not work at all. Is this a bug or

Re: [Asterisk-Users] How to set busy

On Sunday 09 April 2006 06:02, Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. ${DIALSTATUS} BUSY comes from the phone.

Re: [Asterisk-Users] How to set busy

On Sunday 09 April 2006 08:46, Benoit Panizzon wrote: On Sunday 09 April 2006 06:02, Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up