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Asterisk Users Mailing List - Non-Commercial Discussion
December 2006 - page 1
Benjamin Jacob — 116493996801 Dec 2006
Hello ppl, The scenario : I restart asterisk, sip show peers shows nothing. I make a call from 7013 to 7011. I get the following o/p : SIP Seeding pee...
Kevin Boddy — 116495321401 Dec 2006
Hi I've got an 8 port Junghanns ISDN BRI card with 4 ISDN lines connected to it; all 8 channels are configured in my zaptel.conf. I am having a pr...
Peter Vedstesen — 116495412601 Dec 2006
Hey. Maybe this question has been asked before, but I am new here. I would like to use streamed radio station as "music on hold". I have tri...
Artifex Maximus — 116495445601 Dec 2006
Hello, I have this setup: Telco --PRI(g1,ext-incoming)--> Asterisk TE405P --PRI(g2,int-incoming)--> Alcatel OXO extensions.conf: [ext-incoming] ...
H323 NAT Problem (1 Reply)
Jason Kim — 116495877501 Dec 2006*
Hi, I installed asterisk with oh323. My gatekeeper is out of nat device. How can i register * to gatekeeper? Thanks in advance.. Jason....
Doug Crompton — 116496147801 Dec 2006
Anyone that uses the spa3k with Asterisk knows about the dtmf issues of not being able to get tones properly to an IVR after call completion. You can ...
Richard Minshaw — 116496251601 Dec 2006
[ArchiveOrange]: There doesn't seem to be anything here...
omar parihuana — 116496418301 Dec 2006
Hi Guys, I'm implementing my Asterisk step by step, so far the communications between softphones, hardphones with Gateways, voice mail, are workin...
Cisco IAXmodem HylaFAX (2 Replies)
xneptuno — 116496489701 Dec 2006*
Hi, I have a question: In my company a have a 100% IP Telephony implemented (Cisco Phones, Routers, Switches, etc). We are connected to our operator b...
gc — 116496546301 Dec 2006
When I use AgentCallbackLogin() function to login an agent, I got following warning message saying the agent is not valid for auto login while my othe...
app_sql_postgres gone in 1.4 (1 Reply)
Gavin Hamill — 116496587301 Dec 2006*
Hi, I'm putting together a system to manage agents with Realtime, and without chan_agent. In 1.2.13, there's a handy (although marked as depre...
Hall, Eric M. — 116496601101 Dec 2006
Group I have app_swift working on our asterisk server running 1.4-Beta3. My question is can you read variables with it? Like reading back callerid num...
Asterisk: SIP Gateway or Proxy (4 Replies)
yusuf — 116496699401 Dec 2006*
Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway ...
Damon Estep — 116496832901 Dec 2006
I have posted a bounty in the dev mailing list for this issue, if you are similarly impacted feel free to join in. This is a 2 month old issue with si...
Patrick Fortin — 116497384601 Dec 2006
Hi I have the following setup phone - mta - asterisk - patton_sn2400 - PRI I am trying to program *67 to block caller id name and number To do this co...
direct IP calling with extension (1 Reply)
Jerry Geis — 116497490801 Dec 2006*
All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. No...
No caller ID, no incoming call (4 Replies)
jezzzz . — 116497802301 Dec 2006*
Is it possible to reject all incoming calls that do not have a CID? Could I do something like that (modified version from the book): exten => 123,1...
Yuval Yogev — 116497975401 Dec 2006
Hi all, Has anybody a clue how to pass the call immidiatly ? At the MP104 I put at the autodial a number of a sip extention. Also, I noticed that afte...
Bruce Ferrell — 116498359301 Dec 2006
I've found an interesting behavior in callerid handling. I have very long callerid coming in or maybe just improperly combined information. In any...
direct IP calling with extension (1 Reply)
Jerry Geis — 116498447101 Dec 2006*
/>>/ />>/ If I have video phones behind an asterisk server (with 2 network cards) />>/ and all the phones have extensions. Internall...
Jerry Geis — 116498760201 Dec 2006
Presently I have a wav file (voice) for my call attendant. How do I specify a video file for a call attendant for video phones? Thanks, Jerry...
Vincent Delporte — 116499135801 Dec 2006
Thanks. That what was missing. In rtp.conf, I fixed ports 10000-10019 and mapped those ports on the router, and it worked. >Also, Xlite uses (or us...
Damon Estep — 116499138601 Dec 2006
Caller ID should always be either ANI + CNAM (where available) on inbound, or "anonymous" (No ANI). If you are getting anything different fr...
Justin Findlay — 116499368401 Dec 2006
I have an asterisk server that connects to my voip provider over iax2. Some of the POTS phone numbers I've called consistently get no tx audio. Th...
CALL TRANSFER (1 Reply)
Damon Estep — 116499532901 Dec 2006*
Your dial string must have either the t or T option set....
Damon Estep — 116499551801 Dec 2006
Be careful, if you set both T and t you might be allowing the wrong party to transfer the call! In MOST cases you would want T or t, not T and t, alth...
sip address in voicemail emails (3 Replies)
Mark Price — 116504961102 Dec 2006*
hi, I am using asterisk 1.2.10. I am trying to send sip links in asterisk voicemail, so that users can easily reply to emails. This does not seem to b...
Gavin Hamill — 116505257202 Dec 2006
Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings seem to work, I just wanted confirmation that I'm not doing some...
Problem in Poland (1 Reply)
Alex Rixhardson — 116505760302 Dec 2006*
Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How ...
Alex Rixhardson — 116505928802 Dec 2006
If I'm not mistaken (that's how I was told), the inbound calls are managed by Telekomunikacija Polska, and outbound calls are managed by Profu...
Caller ID Rewrite (9 Replies)
David Bath — 116505953102 Dec 2006*
Hi All, I have a quick query which I'm sure someone will have done before. Essentially, I have a 3rd party desktop app which does number lookup in...
Alex Rixhardson — 116505986702 Dec 2006
This is what PRI debug says on problematic call:-- Called g1/482 < Protocol Discriminator: Q.931 (8) len=13 < Call Ref: len= 2 (reference 3/0x3)...
Lars Knopf — 116507216402 Dec 2006
Hi! I am having problems with rxfax. When receiving a fax (on a Zap channel from a te110p), I see on the console: Dec 2 18:49:22 WARNING[31532]: chann...
Watkins, Bradley — 116508978102 Dec 2006
Well, I can't pretend to know how other people use it, but perhaps an example of how I use it would be helpful.Most of the sites that I maintain h...
Nick Adams — 116509975402 Dec 2006
Hello, Can anyone comment on the success of AMD/NVMachineDetect in an Australian setting? What kind of hit/miss ratio can we expect on a good quality ...
Linksys PAP2t-NA and Asterisk (5 Replies)
Jason Michaelson — 116510459203 Dec 2006*
I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Ast...
Alex Rixhardson — 116512051103 Dec 2006*
Hi guys, Here is a bit more detailed information of my problem: If I connect Asterisk PBX to the Polish telco via E1, I don't get any red alarms o...
Dovid B — 116512968203 Dec 2006
Hi List, I just wanted to let eveyone that RNK uses asterisk. One more big company to let clients know that asterisk is great....
Derek Whitten — 116513428103 Dec 2006
Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No powe...
Dovid B — 116513516903 Dec 2006
If I have SIP Provider (sending call as G729) > Asterisk (picks up and asks for PIN) ---> call sent to SIP provider I understand that I will nee...
RTP Media Path (2 Replies)
Dovid B — 116513922103 Dec 2006*
I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider > Asterisk...
Dave Morrow — 116514726003 Dec 2006
I wonder if anyone has experienced an issue I have found with the Linksys SPA-841 phone.On my Asterisk (Trixbox 2), to login to a queue, a user must e...
Peter Braidwood — 116515378803 Dec 2006
Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills the same way as he answers followups ;-) g -Original Message- From: Pet...
Esteban Guana-Jarrin — 116515824103 Dec 2006
Hi List, I am experiencing an issue with a server running asterisk; I installed an AVM FRITZ card and configured it to work with the capi module. Once...
Matt Gibson — 116515841603 Dec 2006
Hi All, To coincide with my Asterisk 1.4 Beta Howto, I've also started the spandsp3 howto. The howto is functioning, however when attempting to us...
Matt Gibson — 116515853303 Dec 2006
A while ago I wrote a numbers guessing game to keep me entertained on those really boring days :). I've uploaded it to the blog for the rest of yo...
Steve Totaro — 116517263603 Dec 2006
Not sure if anyone has posted this before, but it would be great (providing it works well) to connect this to an FXO port. http://www.jr.com/JRProduct...
G729 Passthru? (2 Replies)
Matthew Rubenstein — 116517794003 Dec 2006*
I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are ...
Gidean Chan — 116519524304 Dec 2006
Hi, could anyone tell me how to extend the time for Asterisk to pick up an incoming PSTN call ? Thanks Gidean...
Remco Barendse — 116519563404 Dec 2006
I am using Asterisk 1.2.13 with Zaptel 1.2.11, I used to have an old PBX connected to one port and the PRI connected to the other. I'm having seri...