search for: phone4

...ee.cc patching file src/modules/Text/token.cc patching file src/modules/Text/xxml.cc patching file src/modules/UniSyn_diphone/us_diphone_index.cc __SNIP__ so I am patching them. I setup and extension to test festival and when I dial it I get . __SNIP__ -- Executing Answer("SIP/phone4-17ae", "") in new stack -- Executing Festival("SIP/phone4-17ae", "mary had a little lamb") in new stack == Parsing '/etc/asterisk/festival.conf': == Parsing '/etc/asterisk/festival.conf': Found telco-pbx*CLI> SIOD ERROR: unbound varia...

...ctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing Dial("SIP/phone3-2c9f", "SIP/phone4") in new stack -- Called phone4 -- SIP/phone4-aaf5 is ringing -- SIP/phone4-aaf5 answered SIP/phone3-2c9f -- Attempting native bridge of SIP/phone3-2c9f and SIP/phone4-aaf5 == Spawn extension (sip, 2, 1) exited non-zero on 'SIP/phone3-2c9f' I'm a newbie, so I don...

...uot;Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone3" <3> disallow=all allow=gsm [Phone4] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone4" <4> disallow=all allow=gsm I use the following extensions for asterisk (extensions.conf): [sip] exten => 1,1,Dial(SIP/Phone1,20,tr) exten => 2,1,Dial(SIP/Phone2,20,tr) exten => 3,1,Dial(SIP/Phone3...

Hi! I have two, not too related questions: - the probably simpler one: if anyone can help me out using a Cisco 7905G with chan_sccp? I did already managed to get it working with a SIP image, I'd just like to see it work with this one as well. It's probably something I screw up with the configuration, as the phone registers, only I don't get any lines with it, although I have it

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 69.90.155.70 -- SIP/fwd.pulver.com-f526 is circuit-busy == Everyone is busy/congested at t...

I'm new in hire so Hello to everyone! I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration (used make samples). I would like to make phone connections between X-Lite (SIP) installed on computers in LAN. How to make this? I was reading manual, and tried to make changes in sip.conf but this all

...SDN etc all seems to work ok until either end picks up the phone. The following error occurs: Jan 13 18:32:51 NOTICE[23016]: Unable to find a path from g723 to slin Jan 13 18:32:51 NOTICE[23016]: Unable to find a path from ulaw to g723 Jan 13 18:32:51 WARNING[23016]: No path to translate from SIP/Phone4-cfa4(1) to Modem[i4l]/ttyI1(64) Jan 13 18:32:51 WARNING[23016]: Can't make SIP/Phone4-cfa4 and Modem[i4l]/ttyI1 compatible Jan 13 18:32:51 WARNING[23016]: Bridge failed on channels SIP/Phone4-cfa4 and Modem[i4l]/ttyI1 And the call is dropped........ I'm using the identical config files...

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

...t call each other and I will get message (in * CLI) that particular extension does not exist in a given context Here are my contexts definitions: [from-sip] exten =>101,1,Dial(SIP/phone1) exten =>102,1,Dial(SIP/phone2) exten =>103,1,Dial(SIP/phone3) [iax-user] exten=>201,1,Dial(IAX2/phone4) exten=>202,1,Dial(IAX2/phone5) If I try to call from IAX2 phone to say ext 102, I get "request '102@iax-user' does not exist" I have tried to include iax-user in from-sip and I can make calls from SIP phones to IAX2 ones, but not the other way around. Now for an interestin...

...secret=secret1 auth=md5 host=192.168.3.60 context=incoming trunk=yes qualify=3000 disallow=all allow=ilbc voip-kntl:~# more /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] ; Global Variables ; Internal SIP Phone Numbers PHONE1=SIP/1001 PHONE2=SIP/1002 PHONE3=SIP/1003 PHONE4=SIP/1004 ; Other Site Authentication SITE1=IAX2/user0:pass0@site0 ; MACRO SECTION [macro-callextension] exten => s,1,Dial(${ARG1}) exten => s,2,Hangup exten => s,102,Playtones(busy) exten => s,103,Wait,30 exten => s,104,Hangup [internal-site1] ; Internal Extension Numbers exten =&...

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

...olycom SoundPoint IP 600 SIP phones. We want a web-based login interface for the phone system. Basically, someone will go to a station which has a computer and a phone. They go to asterisk.mycompany.com and are prompted for a login and password. Each phone has a name. Phone1, Phone2, Phone3, Phone4. The first time they login, they should be asked to select a default phone. Once they select their phone and are logged in, Asterisk should route all calls for that users extension to the phone they have logged in to. This will also be used with the Queue system. We also need full reporting...

Hi, I have an FXO card connected to my phone line which works in Asterisk as Zap/1. Is there any way of detecting whether something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

...When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten => s,7,Voicemail(u${PHONE1VM}) exten => s,8,Hangup exten => s,107,Voicemail(b${PHONE1VM}) exten => s,108,Hangup Do you see anything wrong with this ? Regards...Martin -- Anything is possible on paper. -- Ron McAfee

I'm using * at home and I planned on having * let the answering machine in my kitchen to the "general" voicemail getting. However, about 6s into the call * will hang up the line. I found a post about OHT somethingorother, so I can probably work around it, but I'd like to know what's happening and if there's a better way around. Thanks! -- -M There are 10 kinds of

...e this: 1. Few ATA connected to 3 phones + network 2. Linux router that lies between * box and network. Virual interfaces, iptables SNAT/DNAT ..... 3. * properly configured connected to the network. I looked for * configs, but I did not find a way to send traffic for phone1-phone3 as Public_IP_1, phone4-phone6 as Public_IP_2 ...and so on. Anyone ever did something like this using just a box? Any ideas are welcomed. Thank you, Bogdan

This may be obvious but I have not found the answer in the archives or web searching. I am in the process of transitioning to Asterisk. While I have two systems connected to the same PSTN line, I want to configure Asterisk to not answer an incoming call. Is this a setting that you would have in the zapata.conf file? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...s on the phone. UserB-phone 2 shouldn't get any calls. Illustration: (hope it don't get messed up) Incomming Call --> My Company --> Group1 --> UserA-phone1 --> UserA-phone2 --> Group2 ---> UserB-phone3 ---> UserB-phone4 --> Group3 ---> UserC-phone5 ---> UserC-phone6 So what I want per group level is that only one user (phone) is active at the time. And if all of the groups are busy, I want to send the caller to voicemail. Everyone can call out at the same time, but it must update...

...hone1](phone) signalling = fxs_ks callerid = "Andrew F Robinson" <(503)543-2338> dahdichan = 1 [phone2](phone) signalling = fxs_ks callerid = "Michael C Robinson" <(503)987-1322> dahdichan = 2 [phone3](phone) callerid = "2010" <2010> dahdichan = 3 [phone4](phone) callerid = "2011" <2011> dahdichan = 4 [root at robin asterisk]# extensions.conf: [globals] CENTURYLINK=DAHDI/1 COMCAST=DAHDI/2 ANDREWROOM=DAHDI/3 SERVERROOM=DAHDI/4 [external] exten => _9NXXNXXXXXX,1,Dial(${CENTURYLINK}/${EXTEN:1}) exten => _8NXXNXXXXXX,1,Dial($...