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Asterisk Users Mailing List - Non-Commercial Discussion
October 2009 - page 1
chanspy and DISA (7 Replies)
John Millican — 125435931001 Oct 2009*
Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so th...
Moises Silva — 125436285801 Oct 2009*
Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected t...
Choose IAX or SIP trunking? (10 Replies)
Kirill 'Big K' Katsnelson — 125436305801 Oct 2009*
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunki...
Richard Kenner — 125436579601 Oct 2009
My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and t...
das sandesh — 125437301501 Oct 2009*
Hi All, I have a problem, when I was doing a performance testing using an asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls...
randulo — 125439233101 Oct 2009
This week Steve Sokol stops by to describe and field questions about Digium's new affordable speech recognition solution. Later on in the call, we...
RTP Delayed during RTCP (1 Reply)
Cyprus VoIP — 125440184701 Oct 2009*
Hello, Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? Can that be fixed? Thanks....
Busy app timeout (1 Reply)
Julian Lyndon-Smith — 125440195601 Oct 2009*
Using 1.4 svn, I want to implent the busy application. With the following dialplan: [inboundqueue] exten => _X.,1,Answer() exten => _X.,n,Goto(d...
Rennes Neps — 125440544301 Oct 2009*
Hei! Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I'm setting up a custom CDR fields and I was wond...
INVITE Sending Local IP (5 Replies)
Mike Bessette — 125440811001 Oct 2009*
Hello. I set up an Asterisk box a couple days ago and was having problems with not being able to hear SIP clients. After some troubleshooting we have ...
portech MV-378 SIP GSM Gateway (1 Reply)
robert boardman — 125441927401 Oct 2009*
Hi All I having an intermittent problem with the above mobile gateway and would appriciate some advice basically 1 in 10 calls fail at some point duri...
QOS/DSCP for IAX? (3 Replies)
Michelle Dupuis — 125442368101 Oct 2009*
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.confThanks...
IAX2 Call rejected, CallToken Support required (2 Replies)
Klaverstyn, David C — 125445252402 Oct 2009*
Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x whic...
James Lamanna — 125446624702 Oct 2009
Hi, I have two asterisk boxes A&B connected together via IAX. Phones register to Asterisk box A, and Asterisk box B is the PSTN connection. When d...
Mindaugas Kezys — 125447332502 Oct 2009
Hello, We are happy to announce that FREE version of MOR 8 - our advanced Softswitch with billing and Routing is released. It comes as ISO image which...
Szasz Szabolcs — 125447463202 Oct 2009
Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make call...
Zeeshan Zakaria — 125449032302 Oct 2009*
Greetings, I have created simple conferencing solution before using meetme application, but this times its a little tricky. My client needs a function...
Creating a clear channel on zaptel (1 Reply)
Zeeshan Zakaria — 125449372402 Oct 2009*
Hi, Is it possible to create a clear zaptel channel which doesn't require to be picked up? The requirement of my client is to open a clear channel...
Pablo Bernasconi — 125449506202 Oct 2009
Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need i...
Followme (7 Replies)
Anahi Ludueña — 125449723702 Oct 2009*
Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme...
David Cook — 125450516802 Oct 2009*
Has anyone seen something like this before. Randomly, on longish calls, the local side of the call audio goes dead. Meaning remote caller can hear us ...
Alan Lord (News — 125450572302 Oct 2009
Just FYI Really, nothing to do with me... http://www.thevarguy.com/2009/10/01/systems-i......
Extra Sounds Missing on 1.6.1.6 install (3 Replies)
Mark Hulber — 125450995402 Oct 2009*
It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sou....
What are the reasons for VoIP echo? (10 Replies)
Myles Wakeham — 125451240102 Oct 2009*
I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two ...
Softphone in Web (9 Replies)
ABBAS SHAKEEL — 125451519402 Oct 2009*
Hello I am thinking to develop a softphone that is integrated into web.(in form of APPLET or some thing else) Ie a user with with just a PC with Net B...
help on ${RTPAUDIOQOS} (1 Reply)
Asterisk User — 125452185202 Oct 2009*
Hi All, While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in my dialplan. I had 2 sip extensions 555 and 666 and I called from...
Pablo Bernasconi — 125453045703 Oct 2009
Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need i...
Asterisk User — 125454983503 Oct 2009
Klaus, Yes I do have set canreinvite=no in sip.conf. One more thing I noticed is following two cases when I replaced exten => _x.,n,Dial(SIP/666,30...
app_hackblock to prevent SIP/IAX reg trolling (6 Replies)
Michelle Dupuis — 125457979703 Oct 2009*
Has anyone written an app that monitors SIP/IAX registration attempts? A couple of clients are being flooded with SIP registrations (but the source IP...
Restarting of B-channel on span 1 (6 Replies)
Aman Dhally — 125458019903 Oct 2009*
Hi All, Hope you all are fine and good, Today i have found that Mine all PRI Channels are restating after every interval of one hour, and i have searc...
Asterisk on NSLU2 : Grandstream not registering (2 Replies)
jonas kellens — 125458482103 Oct 2009*
Hello there ! I have successfully installed Asterisk on a normal server and on a Linksys WRT610N with DD-WRT running from connected USB-stick. This is...
jonas kellens — 125458573303 Oct 2009
Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --...
Asterisk and Jack (2 Replies)
Fabien COMTE — 125458950303 Oct 2009*
Hi, I want to use asterisk with jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd star...
David @ULC — 125459039403 Oct 2009
Ideasip is down today ?...
x100p card (1 Reply)
Alan Zheng — 125465720704 Oct 2009*
Hi All: I have a x100p card and asterisk installed to my computer, can I use this card and a softphone to call other phone? Thanks...
Vijay Gandhi — 125466727804 Oct 2009
Wanted to update everyone, that IP 64.34.173.199 belong to a company Voxalot, they have hacked our system twice and they don't even care to reply ...
Peculiar error message when using Q-SIG (2 Replies)
Richard Kenner — 125471734705 Oct 2009*
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call from the SV8300, I see: [Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226...
After call into console/dsp hangup hear ringing (3 Replies)
Jerry Geis — 125471757905 Oct 2009*
I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then afte...
Gonzalo Marcote Peña — 125475402505 Oct 2009
I want to know what dahdi_dynamic and dahdi_transcode modules are for. What are they purpose?. I have read in this thread: http://www.mail-archive.com...
AEL problem: bug or feature? (5 Replies)
Klaus Darilion — 125475573605 Oct 2009*
Hi! I have a problem with "jump" in AEL: _+43123456789! => jump +22; +22 => { NoOp(); } -> OK _+43123456789! => jump 22; 22 =...
Vadim Lebedev — 125475639305 Oct 2009
Hello, I'm looking for info about interconnecting asterisk to QSIG GF enabled PABX over PRI . Any information and pointers will be helpful. The ve...
DTMF problem during read() (1 Reply)
Bart Fisher — 125475652205 Oct 2009*
I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not ...
dahdi dies with "No more room in scheduler" (2 Replies)
James Lamanna — 125476111805 Oct 2009*
Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Aske...
Jeff LaCoursiere — 125476129205 Oct 2009
Anyone working on this? Would love to have a "click to talk" that would operate with my Grandstream video phones. j...
Method to downgrade asterisk (2 Replies)
Bart Fisher — 125476854205 Oct 2009*
I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the...
OriginateResponse Event (8 Replies)
Anahi Ludueña — 125477180905 Oct 2009*
Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResp...
Grandstream GXW4024 experience (2 Replies)
Olivier — 125480485006 Oct 2009*
Hi, In this http://thread.gmane.org/gmane.comp.telephony.... from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-da...
ooh323 and h323 (3 Replies)
bilal ghayyad — 125481731706 Oct 2009*
Hi All; To those who used H323, which u advise to use: h323 that come with Asterisk or the ooh323? Why? Anyone can advise if still need gnugk for regi...
d tbsky — 125482109806 Oct 2009
hi: in our country callerid is sent with fsk. but it will sent DTAS(dual tone alerting signal) first, then fsk callerid, then first ring. I search goo...
Thomas Janzen — 125483087706 Oct 2009
Hello, i have a big problem... i want to connect my asterisk server to a lancom 1722 device (ISDN/SIP) Gateway. sip.conf: [general] context=default al...