similar to: Basic questions

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hi! I have been looking for a while for informatoin about how QoS is assured in Asterisk, but I haven't found a thing. Can someone give me some tips about that? Thanks, Best regards, Mireia

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

Hi all! One easy question... I hope someone will answer me. I've installed asterisk with the samples. Somewhere in my network I have an H.323 Gatekeeper. What must I do to make that the gatekeeper talk with Asterisk? And I another little question... with the samples installed asterisk works ok? What must I install to see how it works? I am lost!!!!!!!!!! Please help me! See you. Mireia

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

Hi! I am trying to know well asterisk. For that I would like to know the exact role for each config file. Can someone tell me what is the role of the next ones or a web where I could find this information? That will be very helpful. - alsa.conf - enum.conf - modem.conf - modules.conf - oss.conf: what is oss? - parking.conf: what is parking? - rpt.conf: what is radio repeter? - queues.conf -

Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called 612@fwdpulvercom No path to translate from SIP/fwdpulvercom-dd5a(2) to

I want * to answer the phone when call comes-in. I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a command: mknod /dev/phone0 c 100 0 Though, when I start * I get: Parsing '/etc/asterisk/phone.conf': Found Sep 12 00:18:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' Sep 12 00:18:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

Hi, I'm having problems with setting up my ISA LineJack card on Linux machine... I've done everithing according to documentation available, by compiling the new ixj driver (v1.2.1), loading it, adding device node /dev/phone0 with major number 100 and minor number 0, adding aliases into the /etc/modules.conf: alias char-major-100 phonedev alias char-major-100-0 ixj and running depmod

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Greetings: We have been running a CVS HEAD version of asterisk from Mar 10, 2005 on ix86 (PIII-600) Linux 2.4.27 with ixj (chan_phone) hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER