search for: phonelin

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. I would appreciate any info that yo...

Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very appreciated. I'm and absolute newbie on asterisk, btw. Thanx! Kosa - Un mundo mejor es posible -

...for a while outside Australia, is there an SMS module for Asterisk which would make use of it? I think that being able to receive (and probably send - haven't even started looking at that yet but it is supported in the same way) SMS messages would be a really nifty thing to be able to do from a phoneline, and would save me buying a $600 GSM modem to do the same thing! Thanks James

...er hangs up, asterisk treats it as a new call. I have tested this with asterisk from cvs HEAD as of today, and the "stable 1.0 version". zaptel driver is cvs head. Also as of today. I have also tested with both callprogress = yes and no, but no luck. I plugged in my multimeter into the phoneline to see what actually happens here. This is how a call is signalled in sweden: 1) Line polarity reverses This is to mark the beginning of the CallerID 2) CallerID is sent using DTMF Don't know the system used, but I remember seeing a post about it in the archives. 2) Polarity reversed...

I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was unsucessful in finding anything on google and wiki about it. jacob

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

Hello, I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it will not pick up an incoming call. Any suggestions/tools to see what the problem is? I have looked at zttool where this line changes but I don't understand what it means (The last digit changed from 0 to 1) Total/Conf/Act: 4/ 1/ 1 /etc/zaptel.conf fxsks=4 loadzone=nl defaultzone=nl

...y at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and their sip server. (at the moment for free) If you had the possibility to connect asterisk as a phone to this server it would be an easy (and cheap!) way to realise a gateway to old-style-phoneline. Waiting for reply, Birk Bremer -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAGg6+7QhrwFQeHVsRAo+1AJ9+gk79nIxbxt6rPPpHIBw2MZibBQCdEcJN wWawRjIjmpUs9orqrmEEcNI= =EGhT -----END PGP SIGNATURE-----

...at is... >I installed an Asterisk@home machineand configured a few SIP accounts on >it. They seem to run fine inside my network, so that's OK. Now, I want to >start using a X100P to connect it to my phone line, to make call routing >between internal SIP phones/softphones, my local phoneline and an external >SIP server. How do I enable and configure the X100P? >I ran the configuration tool locally on the machine (the genzaptelconf >thing) and it added a line to the config. >Now using the number it gave me, in the trunk config in AMP, I still cannot >get an outside line...

...hat phonenumber, and sometimes its complete. My personal guess its something with the timing or so, but no clue exactly where to look on that part. The versions im using: Asterisk 1.6.1.12 Dahdi 2.2.0.2 \-- Echo Cancelling engine: MG2 Maybe someone has an idea, i really dont atm. Its btw a dutch phoneline. Regards, Evert

...no function existed to do that, what I could find. Anyone knows about one? My second idea, was to try calling simply, to know if things were ok. But, I couldn't just call any number, I had to know the number was in use, and not disturbing anyone. So, I called myself, or I called another of my phonelines. So, I'd like to use the asterisk manager interface in java to originate a call from one ZAP-channel, calling out to my telephone provider, And then they will direct the call back to my, but into another ZAP-channel (since I'm calling that channel's number). So: I'm making ZAP/1...

...PCM cards.pcm.hdmi ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave Cannot connect to server socket err = No such file or directory Ca...

...with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I connect an ordinary phone to it the led on the phone flashes once when someone hangs up on me. Also, if the person that calls me hangs up AFTER me, asterisk seems to interpret that as another incoming call. Also, sometimes I get these error-messages: WARNING[29711]: Ring/Off-hook i...

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number is really sent on the PSTN by connecting a number presentation box between. I see the incoming number when I dial into asterisk. But Asterisk never gets any number. I tried disconnect the num...

Alarms Span RED wildcard X101P Board1 OK wcusb/0 0 ive got my asterisk server up and running and working correctly, the first time after a reinstall and reboot everything was fine - i had both alarms OK and i could get the USB extension ringing when i ran the house number from my mobile. as soon as i tried again i got a red alarm on the wildcard board. now im using the sample

...Been trying to send this message for 2 days now, without success so far... --- Hello, I'm exploring the capablities of Asterisk and must say I'm really impressed! However, I don't need most of the options, but can't figure out the things that appear simple to me... We share our 3 phonelines with 20 users (students). There's a (pre-WWII) ISDN-PBX which some of the students want to keep. Since only 5 handsets can be attached I thought Asterisk would be a perfect solution for the remaining 15 students. So far I managed to get things running, but if I configure extensions.conf like...

Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at

...ll do up to 5Mb/s down and 256Kb up. I'm not sure if it's full or half duplex. But it sounds gread. At this office where I normally work, I have a wireless broadband connection to a POP in a town 5mi from here which connects via wireless to a town 10mi from there and then it's either phoneline or wireless to the ISP's ISP. It seems like it might be a dicey connection, but I've had tremendous luck with it for over 3 years now. As a matter of fact, I regularly use a MultiTech MultiVOIP with a 9.6kb/s voice coder on it and hardly anyone notices that it's a VOIP connection. (...

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3