[asterisk-users] asterisk-users archive

http://lists.digium.com/pipermail/asterisk-users/ Trusting user-generated date fields? sweet. :D -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asteris

Re: [asterisk-users] app_swift 1.6.2 DTMF issue

On 1/10/2010 5:33 PM, Jeremy Kister wrote: > With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you > enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} > is [un]set in an odd way. I fixed it up by ignoring the f->subclass and starting the dtmf_listener right away.

[asterisk-users] PHP-Script (AGI) doesn't finish after upgrading to 1.6.0.15

Hi, I recently upgraded our asterisk server from some 1.4 version to version 1.6.0.15. From this point on my AGI scripts aren't working anymore, here is a simple example: [isdin] exten => 83086921,1,AGI(test.php) exten => 83086921,2,NOOP("MARKE1") exten => 83086921,3,WAIT(2) exten => 83086921,4

Re: [asterisk-users] app_swift 1.6.2 DTMF issue

On 1/10/2010 5:33 PM, Jeremy Kister wrote: > With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you > enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} > is [un]set in an odd way. The problem lies within f->subclass inside the else if of line 436. the code seems to

Re: [asterisk-users] You won't help me anymore?

On Mon, Jan 11, 2010 at 6:23 AM, Zhang Shukun wrote: > you'd better paste your dialplan snip here, in order to get specific help. > > 2010/1/11 Darrick Hartman : > > On 01/10/2010 11:38 PM, hadi motamedi wrote: > >> > >> FWIW, he did post his question yesterday. I've just taken a look and >

Re: [asterisk-users] You won't help me anymore?

you'd better paste your dialplan snip here, in order to get specific help. 2010/1/11 Darrick Hartman : > On 01/10/2010 11:38 PM, hadi motamedi wrote: >> >>     FWIW, he did post his question yesterday. I've just taken a look and >>     one potential issue I've spotted is that the external server h

Re: [asterisk-users] You won't help me anymore?

in your dialplan ,did you add area code automaticly? when dial out. 2010/1/11 hadi motamedi : > > > On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane wrote: >> >> On Sunday, January 10, 2010, Francesco Peeters wrote: >> >> > Yes, post your question clear and consicely, include all relevant >> > informa

Re: [asterisk-users] You won't help me anymore?

On 01/10/2010 11:38 PM, hadi motamedi wrote: > > FWIW, he did post his question yesterday. I've just taken a look and > one potential issue I've spotted is that the external server he > mentions is 192.168.0.139, which is part of the 192.168.0.0/16 > > ne

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

At 15:33 1/7/2010, Tzafrir Cohen wrote: >On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote: >> At 00:22 1/7/2010, Tzafrir Cohen wrote: >> >On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote: >> >> At 16:49 1/5/2010, Tzafrir Cohen wrote: >> >> >On Tue, Jan 05, 2010 at 04:24:37PM -0600,

Re: [asterisk-users] You won't help me anymore?

On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane wrote: > On Sunday, January 10, 2010, Francesco Peeters wrote: > > > Yes, post your question clear and consicely, include all relevant > > information and snip all unneccessary history. > > > Note that: no reply != not wanting to help... > > It *is* obv

[asterisk-users] How to test if a telephone is busy now?

hi, all i want to test if a telephone is busy now in agi php script? Could you tell me how to do that judgement？ example： if( ivan is not busy) { $agi -> exec_dial("SIP","ivan"); } else if (test is not busy) { $agi -> exec_dial("SIP","test"); } Thanks very much! -- Best regards, Su

[asterisk-users] How to use AGI php script function $agi -> exec_dial

hi, i want to use $agi -> exec_dial() to dial . this is in extention.conf [tutorial] exten => 1234,1,Dial(SIP/ivan) is that i use $agi -> exec_dial("SIP","tutorial|1234|1") can dial ? BTW, i want to know some turorial on how to use PHPAGI funtions? can you tell me some? Thanks! -- Best r

[asterisk-users] Zhang Shukun 想跟您聊天

--- Zhang Shukun希望通过 Google 的一些最炫的新产品与您保持更密切的联系。 如果您已经拥有 Gmail 或 Google Talk，请访问： http://mail.google.com/mail/b-1f731adb8c-861c77d1cc-d68ecfc46b4cc7de 您需要点击此链接才能与Zhang Shukun聊天。 要获取 Gmail（Google 提供的免费电子邮件帐户，存储空间超过 2,800 MB）并与Zhan

Re: [asterisk-users] Problem with my dialplan

The 8 probably comes from the T1, does the telephone number end with an 8? The playback of ss-noservice might be a fallback ensuring that *something* happens when a call comes in On Sun, Jan 10, 2010 at 1:31 PM, Edwin Quijada wrote: > Hi! > I have an T1 line for using with IVR AGI. I receive the

Re: [asterisk-users] Weird Polycom SP 650

Bon journo Aldo. > I am having several issues with my first SP 650. > * Assembly: 2345-12600-001 Rev.G > I have deployed more than 200 IP650 with the same assembly as yours and so far there are no problems. > The first thing I have noticed is that I was not able to upgrade the > unit's

[asterisk-users] Weird Polycom SP 650

Hi, I am seeking help with the installation of a Soundpoint 650 desk phone. Although I have some experience (and a good one! no single issue so far, besides the problem I am trying to solve...) installing a few SP 320/330 units, I am having several issues with my first SP 650. Polycom SP 6

Re: [asterisk-users] Grandstream GXW-4024

On Sun, Jan 10, 2010 at 1:17 PM, C F wrote: > Anyone using the above mentioned SIP Gateway made by grandstream? > I would like to hear some feedback on real life experience using this gateway. I have a few that I used for about 2 days before I replaced them with AudioCodes MP-124s. They worked f

[asterisk-users] app_swift 1.6.2 DTMF issue

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while

[asterisk-users] Problem with my dialplan

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help

[asterisk-users] Grandstream GXW-4024

Anyone using the above mentioned SIP Gateway made by grandstream? I would like to hear some feedback on real life experience using this gateway. TIA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ast

Re: [asterisk-users] Please remove me from the mailing list.

Kevin P. Fleming schrieb: > Rick Green wrote: >> 'dash dash space '. A compliant MUA will strip that >> line and everything after it when quoting for a reply or forward. Note >> for the list admin: Please preceed your message-footer with a sigdashes >> line! > > Good idea, done! A big than

Re: [asterisk-users] Really Silly Question From Total Newbie

Alternatively, since your iphone will have a web browser you could possibly meet your needs going a web development route instead. Asterisk and telephony is a very interesting project though. On Thu, Jan 7, 2010 at 11:20 AM, UIT DEVELOPMENT wrote: > Hello Tiago, I think that this is the route I

[asterisk-users] Directory and Voicemail Problems after upgrading from 1.4 to 1.6

Hello all, I've noticed a few differences in my recent upgrade from 1.4 to 1.6.2 that have me baffled. I thought I'd write to the list and see if anyone has any ideas. - In 1.4, app Directory matched users based on the name listed in their voicemail.conf entry. Now it appears that 1.6 matches o

[asterisk-users] Off-line subscribed phone amber on SPA942?

If xlite subscribes on a hint, and the phone is offline, xlite says so ("not online") If SPA942 does the same, the led is green for "available". The other hints work: blink red for ringing and red for busy. I seem to remember the led once showed amber for subscribed phone offline. The SPA exten

Re: [asterisk-users] You won't help me anymore?

On Sunday, January 10, 2010, Francesco Peeters wrote: > Yes, post your question clear and consicely, include all relevant > information and snip all unneccessary history. > Note that: no reply != not wanting to help... > It *is* obviously possible people just do not KNOW the answer!... (Oh > what

Re: [asterisk-users] You won't help me anymore?

ABBAS SHAKEEL wrote: > why don't you post your question > > On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi > wrote: > > > > On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra > wrote: > > Sunday, January 10, 2010, 11:24:22 AM, hadi

Re: [asterisk-users] You won't help me anymore?

why don't you post your question On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi wrote: > > > On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra wrote: > >> Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: >> >> > You are not willing to help me anymore ? >> >> Why do you think this? >> >> -- >> Be

Re: [asterisk-users] You won't help me anymore?

On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra wrote: > Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: > > > You are not willing to help me anymore ? > > Why do you think this? > > -- > Best regards, > Gergomailto:csi...@gmail.com > > > -- > __

Re: [asterisk-users] You won't help me anymore?

Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: > You are not willing to help me anymore ? Why do you think this? -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation

[asterisk-users] You won't help me anymore?

Dear All You are not willing to help me anymore ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/l

Re: [asterisk-users] No dial-tone with X101P FXO card

On Sun, Jan 10, 2010 at 1:48 AM, Tzafrir Cohen wrote: > On Sun, Jan 10, 2010 at 12:49:24AM -0800, Nitin Bahadur wrote: > > | What's the output of: > > > > > > > >> dialplan show internal > > >> > > >> in the asterisk CLI? > > >> > > >> NB> > > > jserver*CLI> dialplan show internal > > > [ Context

Re: [asterisk-users] No dial-tone with X101P FXO card

On Sun, Jan 10, 2010 at 12:49:24AM -0800, Nitin Bahadur wrote: > | What's the output of: > > > > >> dialplan show internal > >> > >> in the asterisk CLI? > >> > >> NB> > > jserver*CLI> dialplan show internal > > [ Context 'internal' created by 'pbx_config' ] > > '_X.' => 1. Dial(Zap/1)

Re: [asterisk-users] No dial-tone with X101P FXO card

>Some more background...I have a comcast phone line > > which I have connected to my FXO port. When I call my > > number, it goes directly to comcast voicemailin other words, > > there is no ringing tone and pickup by asterisk. > > That would suggest the card is looping the line (busying i

Re: [asterisk-users] No dial-tone with X101P FXO card

On Sun, 2010-01-10 at 00:25 -0800, Nitin Bahadur wrote: > Hi Tzafrir, > >Some more background...I have a comcast phone line > which I have connected to my FXO port. When I call my > number, it goes directly to comcast voicemailin other words, > there is no ringing tone and pickup by asteri

Re: [asterisk-users] No dial-tone with X101P FXO card

| What's the output of: > >> dialplan show internal >> >> in the asterisk CLI? >> >> NB> > jserver*CLI> dialplan show internal > [ Context 'internal' created by 'pbx_config' ] > '_X.' => 1. Dial(Zap/1) > [pbx_config] > > -= 1 extension (1 priority) in 1 context. =- > jserver*CLI> dial

Re: [asterisk-users] No dial-tone with X101P FXO card

On Sun, Jan 10, 2010 at 12:25:10AM -0800, Nitin Bahadur wrote: > Hi Tzafrir, > >Some more background...I have a comcast phone line > which I have connected to my FXO port. When I call my > number, it goes directly to comcast voicemailin other words, > there is no ringing tone and pickup by

Re: [asterisk-users] No dial-tone with X101P FXO card

Hi Tzafrir, Some more background...I have a comcast phone line which I have connected to my FXO port. When I call my number, it goes directly to comcast voicemailin other words, there is no ringing tone and pickup by asterisk. See inline below with NB>... On Sat, Jan 9, 2010 at 11:56