similar to: Disconnect after 12 seconds w/Cisco 303g Phones

Hello, I have scoured google for the last couple of days, implemented some changes but my issue is still occuring. My company uses a hardware Bridge System for conferencing. Typically, users will call in from cell phones but three always call from the VoIP system. Once or twice a day, one of the VoIP phones will just drop. Subsequently, we will hear a modem like sound through the bridge system

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.

I am currently testing an asterisk server with some cisco 7960 phones. I have been having problems with phones rebootin using 6.3 firmware in the asterisk voicemail menus. The phones reboot after a dozen or so random button presses while in the voicemail menus. To try and fix this, I upgraded to sip 7.1 only to find that now the phones reboot even if i'm trying to press a button to

Hi All I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. All other phones don't seem to have it present. Sadly the customer is Quite attached to the Zoiper. I have set QOS =

Hi there! Has anyone succesfully used a Cisco 12 SP+ with *? If so, how did you do? I'v not even tried, but before trying I thought I could bug you somewhat. =) //Filip

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

I've got lines coming in from a legacy system (into FXO ports) which does not give any disconnect notification. Folks familiar with the system say that I can buy or build a device which will listen for so many seconds of dead air and then automaticly send a disconnect signal to free up any hung channels. This seems like something that could be done in software with Asterisk? Right now

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

AAH 1.3 with Cisco 7960 phones/ SIP 7.5 software mostly works great, but there is a problem with one of the phones I use most: It disconnects calls if I dial on speakerphone and then pick up the handset after the other side answers. Thanks in advance for any clues on this. And apologies if this isn't the right forum but I don't know of a Cisco 7960 list. dn

Hello all, I'm new to the list, but use VoIP and * for a little while now. Running Asterisk 1.0.2 on debian linux I'm facing the following problem: I've got two Fritz!Box Fon Adapters (kind of ATA's) with two hardware phone connectors each. So I'm trying to set up a PBX with four internal (SIP) phones. One box has fon1+2, the other fon3+4. When I start up *, everything

Hi all, I've got a user with a couple of Cisco SPA303's. When I dial their phones with a dial string like: dial(sip/phone-a,300,rwkxttT) The phone rings, as expected. However after exactly 60 seconds, I get: [Aug 21 02:09:56] -- Got SIP response 480 "Temporarily not available" back from a.b.c.d:5062 [Aug 21 02:09:56] -- SIP/phone-a-00006a9d is circuit-busy [Aug 21

Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message I?m getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission

I like to think of myself as reasonably intelligent when it comes to systems (MS in Comp Sci, albeit a few hundred moons ago), but I seem to be missing something when it comes to installing NUT on a 64 bit windows XP system. I downloaded and attempted it install but the installer said I needed to install libusb manually. So I went and grabbed that, unpacked the ZIP file and ran the

Hi All; In cisco IP Phone, I can assign for a features for the buttons at the Phone (one button to be for call pickup and one button to be for call forward and button to be for bridge and one button to be for another extension). How can I do the same thin in the Cisco IP Phones (or other IP Phones) while they are registering in Asterisk? Regards Bilal

Hi All; How to configure the buttons in the Cisco IP Phones to be used for different functionalities like "Call Forward, Call Pickup, ... etc"? For example, if I need to assign one of the buttons existed at Cisco IP Phone to be used for CallFrw, how to do this in Asterisk? Regards Bilal

Hello everybody, I would like to use the SpDep-package (especially the Local Moran index analysis and the Getis-Ord statistics) in R for analysing my data. However, I don't have x-y coordinates, but my data is in a distance matrix format. Is it possible to use the SpDep package with predefined distances as well instead of letting the program determining the distance itself by the function

Hi, I heard some time ago that, when running a SIP firmware, Cisco hardphones needed a Cisco call manager to get localized (ie non-english) menus ? Is it still true ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/fcc22a7b/attachment.htm

Hi to all, I'm using asterisk 1.4.9 with chan_h323. When someone in the H323-VoIP cloud dial 1234 this number is assigned to my asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk though the dialplan can delivery the call to a particular SIP phone...this is ok... I can also dial from my sip phone every phone in the H323-VoIP cloud like siemens....BUT...when I call to a

All other phones we work with will auto-answer when we do this: [macro-paging1way] exten => s,1,SIPAddHeader(Call-Info: answer-after=0) exten => s,n,Page(${PAGINGLIST}) exten => s,n, Hangup The SPA phones simply ring. I have verified that Auto Answer Page is set to yes (the default). We've tried a variety of firmware versions and phone ages, going back to an old 942 and new 504s.