similar to: congestion problem with only one number

Hi all, I have a Cisco 7905g with the default cm firmware. It is connected to * using chan_sccp.so. Normal operations with this phone work perfect. Now I'm implementing a click to call app in php. This php app connects to the manager interface and sends the following down the line: $this->_write($fp, "Action: Originate"); $this->_write($fp, "Channel:

hi ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

Hi all I've only been working with Asterisk for a matter of days but have already grown into a big fan =) Much as I've managed to get internal calling working fine, I have a configuration running on an old PII-233 on RH9 with a (although not badged as is a) Dynalink IS64PH/Winbond W6692 PCI Card as /dev/ttyI0. The card works fine in minitel and dials out without a problem.. However try

All, Currently I've got my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones or softphones within my LAN or remote LAN's via VPN. The next step for me is connecting it to the PSTN network. After some tweaking with the modem.conf I got the i4l driver running correctly, and it appears that my Fritz! ISDN v2 card is working correctly. I have added the

I have installed a Billion ISDN card in my Asterisk. Calls between sip and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten => s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten => 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814

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Well, it has been a long time since I used my NetJet S cards with asterisk (moved to the quad PRI card) but I am trying to get this working again for home use. Basically, what I am using is this: Linux 2.6.8.1: config lines: # ISDN subsystem CONFIG_ISDN=y # Old ISDN4Linux CONFIG_ISDN_I4L=y # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y # CONFIG_ISDN_TTY_FAX is not set CONFIG_ISDN_DRV_HISAX=y

Hello! Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN. Yesterday i configured asterisk to be able to hear a song (with Playback() command) with iaxcomm, and it was wonderful. Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card) with asterisk. i configured modem.conf this way, after reading

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

We have setup an asterisk box to let everybody call into the university internal network, but I get unexpected hangups when doing an outbound call from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the call. ----------the dial and the problem----------- -- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new stack --

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>

Hi all, this is a rather "newbie-oriented" question, so please bear with me... The system running Asterisk has been provided with an AVM FRITZ!Card PnP. SuSE Linux 9.0 recognizes it right after booting the system and it seems to be configured (MSN) correctly... The hwinfo looks like this: --- pbx:/etc/asterisk # hwinfo --isapnp 11: ISA(PnP) 01.0: 10300 ISDN Adapter [Created at

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

I can't find solution anywhere. I googled and find people with the same problem but there was no answers on how to fix this. I have W6692 based PCI cards that uses hisax driver (card type=36). Card is working fine under asterisk with i4l modem driver for incoming calls. If I want to dial out using some sip phone I don't get ring indication. Phone is ringing and I hear only silence until

Hey All, I'm still (since April) having problems getting RxFax to work over an ISDN4Linux channel. Just wondering if anyone has had any luck getting it to work? I have done a CVS update today (about half hour ago) and made sure I have the latest version of spandsp according to Steve's website (spandsp-0.0.1k). When I was compiling asterisk, I got the following warnings: