similar to: [Asterisk-User] Howto get the Caller Phonenumber ?

Hi Peoples, Great day for the callingcard-fan ! Just a little mail to let you know that a new version of A2Billing 1.1 (Asterisk2Billing) is available! Many features have been added, lot of bugs solved and hundreds of good improvement made, so there we go -> http://www.asterisk2billing.org The key newest features : * Ecommerce product with API addons - Integration with OsCommerce *

Hello! I finally found a working solution. calling divactrl with the parameter -n [0..20] gives the DID-length means, if you wanna have 123-XXX in digit-wise mode, then call divactrl load -c 1 -n 3 -f ETSI and the card will wait for n digits. regards, Sebastian -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hi everybody, Now I'm using SJphone on a win2k client an * as proxy SIP and GW to PSTN. I have doing some test, but I have the following question. It's possibles to make calls to external PSTN numbers without define an extension to make the call???? I will try to explain-me better. I have done some calls like sip:xisco@A.B.C.D, where in extensions.conf there are an extension like this:

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation

A site I administer has dual T1''s. We have this set up according to NANO-HOWTO[1] using Julian''s routes patch on a late model 2.4 kernel. All is well, has been working well for a long time. Recently one of our T1 providers went belly up and we got something better. Now we have a /28 where previously we only used one IP per interface. Present interfaces: eth0 - T1#1, single

Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general] port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300] type=friend insecure=yes host=xx.xx.xx.xx

Hi All When I dial a number it reaches the asterisk switch as abc at xyz@123.com what I need to do is to parse the abc and send it to my pstn gateway as in exten => _.,2,Dial(SIP/${EXTEN}@pstn.gw) this does work but I do have a varying number of numbers before the @ exten => _.,1,Dial(SIP/${EXTEN:0:12}@pstn.gw) Well can I use some kind of regular expression to take all numbers before

Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there

In a ADS(Adv Serv 2003) setup with a few linux members, I'd like to achieve consistent UIDs for domain users across these linux machines, and idmap_rid seems to be exactly what I'm looking for. However, I cannot get winbind to create uids or gids from SIDs at all. Any hints? --Erik S. Johansen ares samba # smbd -V Version 3.0.10 ares samba # pwd /var/cache/samba ares samba # rm *.tdb

Last month I saw something funny which I can't reproduce anymore: A 0500 number in .au is a service phone number and are forwarded on exchange level to a real phonenumber. So if A calls B it gets forwarded to C. Very simple. Now the funny thing, on the phone of C, I saw both A and B as the "caller id". I've been asking around and trying to get it again with a private 0500

Hi folks! Our company are going to buy an E1 line with Euro ISDN and 30 lines (channels). This is how it will be configured: 3 Lines, of the total of 30, is going to be for the company phones, and share one phonenumber (eg. 555-12340). 1 Line will be dedicated to a specific unique phonenumber (Fax) (eg. 555-54321). The rest of the lines/channels (26) will be used by (by, not for) our customers,

Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The

Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php:

Hi, sorry for this noop question, but does anybody know how to cut the first digit of a variable? example: 044612345 should be after cut operation: 44612345 My try in the extension.conf: exten => _0[0-9].,2,Cut(mynum=EXTEN,/ ,1) exten => _0[0-9].,3,Dial(Zap/g1/${mynum},90,T) but it didn't work, my problem is the delemiter, I have no delemiter, the default is "-" but how to

Hi, I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2). I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I

Hello, How to take the values of channel variables like 'agi_uniqueid' and 'agi_callerid' in agi script. For example #!/bin/bash -x T="$agi_uniqueid" I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script -------------- next part -------------- An HTML attachment was scrubbed... URL: