similar to: alphabetical extension patterns

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk

Hello All, Is CALLERID() setting broken in 1.4.4? My small dialplan : [testclid] exten => _0.,1,Set(CALLERID(all)=Ben Jacob <988077>) exten => _0.,n,Dial(SIP/${EXTEN}) Correct me if I am wrong, Set(CALLERID(all) above supposed to change the display name as above(Ben Jacob) and change the From URI to 988077 at myip?? As of now, only the _display name_ is being replaced, but not the

Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555. Now, when someone calls 1000, and leaves a

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

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Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

Hello ppl, Have implemented a really nice Billing engine using AGI scripts. So far it works fine, tho haven't yet put it in the torture cell. The AGI scripts have been written in PHP, using MySQL for the billing and profile information. The major disadvantages I see using AGI scripts : 1. A new process(invocation of PHP scripts) on every new call. 2. MySQL connections on every instance of

Hello ppl, Searched all over, but couldn't find anything conclusive. Does an off-the-shelf version of Asterisk run without any issues on a 64-bit machine? Does anyone have any 'conclusive' figures? Apologies if this is a repeat question. Would appreciate if I could be redirected to the appropriate link. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it

Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential

All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. Now for people that want to call my video phones external to my office is there a way to do that? On the extenal persons phone enter an IP/EXTEN where IP is my server and not the phone? Can that work? Would I have to have PUBLIC IP address for every

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Regards Mattias -- Mattias Andersson -------------------------------- Storskiftesv?gen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38

Hello :) I have a problem with the Net::HTTP library... The Net::HTTP library [1] uses a Net::HTTPResponse object for all it''s responses from web servers. This class has many subclasses, such as HTTPSuccess, HTTPRedirecttion, etc. When obtaining a response, the library suggests to check what it is by testing the class of the returned object - using case/when or kind_of? (which it does

Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device --------------------------------- Ready

Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? Rgd satish patel

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

Hi, I have switched a while back from chan_capi to chan_misdn. When the number is dialed and the phone is then picked up everything works just fine. Some users however FIRST pick up the phone and then start to dial... I did not get this to work with misdn. When two digits have been dialed, asterisk sees the extension as complete and does not wait for further digits. I am using an midsn NT

Hi group, I'm trying to configure the email notification when a user leave a voicemail, but don't work (send email notification). I configured esmtp in my linux box, if a try to use it with command line, it works fine. (echo "Hello" | sendmail a@b.com -f b@c.com). My voicemail.conf [general] format=wav49 attach=yes serveremail=anonymous@abc.com fromstring=Asterisk

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the