similar to: app_sms woes

Hi, I have been using app_sms for a few weeks now, since I recently upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms doesn't seem to work that well anymore.. On receiving an sms, I execute the app_sms script, and get this as output: -- Accepting voice call from '171701' to 'ournumber' on channel 0/1, span 2 -- Executing Goto("Zap/4-1",

Ok, I just thought I'd publicly pat Adrian Kennard (revk) on the back for this application. This is an excellent contribution and gets my vote for app of the year. For those that aren't aware app_sms allows you to send/receive fixed line sms messages from asterisk. ( you can take a look at a quick page showing this http://www.automated.it/asterisk/sms.html ).. I should point out that

Hi, i've been discovering app_sms and it states that it can act as an smsc for landline sms. Receiving SMS from my Gigaset Phones is no problem and the SMS are stored as files on by * box. So far so good. Let us assume that i have a couple of phones which should be able to receive SMS directly from my * box ( and not from an SMSC from BT or Deutsche Telekom ), So all these phones have

Hello, i try to send out a sms, but with no success. The trunk is a E100P, and the sms should go out to the Telekom SM-SC. What i want to to at the first run is, sending out a sms when a certain number is dialed. I tried: In extensions.conf: exten => 35953,1,SMS(${TRUNK}/9350193010,,0179NUMBER,"Hi there") exten => 35953,2,SMS(${TRUNK}/9350193010) exten => 35953,3,Hangup

Hi, i've been trying to get fixed line sms working for some time now. Can anybody tell me, if he is actualy using this with asterisk in germany? I have followed the instructions found on voip-info. I was successful a couple of years ago with asterisk 1.0.7 and an normal telekom isdn line. Now i want fixed line sms over an Dokom PRI with Asterisk 1.2.9. Here in Germany the Materna

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

Anyone here that knows some hardware USB "client" (DECT) that can connect to a DECT base station and act as a "provider" in asterisk? Ergo, asterisk is the DECT handset, and connects to the DECT base station. Its also important that it also works with most asterisk-compatible devices, including for example raspberry's. The reason I ask, is that many providers in

I've received my Com-on-Air DECT card this morning and while I have configured it (both the DECT registration and SIP to Asterisk configuration), I am running into a "Call missing Call ID from" error message. I can call from a DECT handset to an extension, but not the reverse. Not being an expert, it seems to me that Asterisk is looking for a CallerID from the DECT card. I

I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but

Hello, I need to setup this configuration: - asterisk as IVR; - dect phones. So basically I need a "standard set" of features: - each dect phone has its extension so I can call it directly; - handover of a call with "R" key; - if a call is not replied by someone ring all phones. I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip base station. Which

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

And a PRI/BRI gateway? Is there any product ? Thanks Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Peer Oliver Schmidt Inviato: venerd? 20 gennaio 2006 10.17 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Dect to SIP PCI card Giordano Grandis wrote: >

Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

Hi, It seems to me that DECT Gigasets do not support MWI when connected to a Polycom/Kirk DECT base station : when a new message is dropped into user's mailbox, I can see a NOTIFY message sent by Asterisk to Polycom/Kirk DECT base station (here a KWS300) but Gigaset's handset MWI remains unlit. At the same time, a Polycom/Kirk handset subscribed to the same base station would turn its MWI

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT

Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so I can handle all DECT phones centralized and plug it inside asterisk ? Thanks, Peter