[Asterisk-Users] Do Not Disturb
We have just finished installing some Sayson 480i phones ( will post a review soon) and I have one issue. I cannot seem to get the *78 , *79 ( and like) functions to work. Are these automatically installed with Asterisk? Anything required in the extensions.conf or sip.conf? When I dial *78 I get a 404 error on the phone ( Call failed). Nothing shows in the Asterisk console. Thanks in advance, Shawn Dillon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Do Not Disturb
I cannot get the Do Not disturb function to work with my Asterisk box. If I dial *78 , or *79 on my phone ( Sayson 480i SIP IP Phone) and I get a call failed message. Is the *78 , *79 function installed by default? Does it use a certain module in Asterisk? Any advice/comments? Shawn Dillon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Second TDM400 card
We just received and installed the second TDM card for our asterisk box. It is installed and gets all the green lights. As well my Debian box lists the modules as found. I have placed a fxsks=1-8 in my /etc/zaptel.conf . In my /etc/asterisk/Zapata.conf if I try to change my channel = 1-4 to channel=1-8 I get errors that it cannot init channel #5. I must be missing something simple. TIA Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP Phone Suggestions
We are in the final stage of a rollout of Asterisk in our company. We had some Polycom IP 600 , a Snom 220 , a Grandstream 102 and recently a Sayson 480i phone. I am interested in anyones opinions in the phone they suggest to implement. I must admit I am a little partial to the Sayson 480i , but if there are convincing arguments with regards to other models I would like to hear them. If anyone has had more experience with the Sayson please let me know. There is a company in Vancouver that deals in them , call NetVoice. As a newbie in the market , they ( George) gave great service and advice. Even called me to see how the Snom 220 was working out ( Great customer service!!). Anyways , your feedback is appreciated. Shawn___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentCallBackLogin
I have the AgentCallBackLogin working well when the support technician logs into the queue manually. If there a way to get certain extensions to automatically log into the queue? That way I do not have to worry about help desk staff forgetting to log into the support queue and never receiving support calls.As far as I know, it's enought to add those extensions as a members inqueue.conf.-- Graf0 Ok, I will share more details with my particular installation. 1) In my extensions.conf I have the following; exten = 997,1,AgentCallBackLogin(999|[EMAIL PROTECTED]) 2) In my agents.conf I have the following group=1 agent = 999,1234,Test Agent 3) In my queues.conf I have he following member = Agent/999 When an agent dials extension 997 and enters their password they will then be included in the support queue for calls. If they do not call extension 997 and enter their password when a call is placed into the support queue a message appears on the console stating that no one is answering the support queue. Is there a way to get a certain extension to automatically log into a support queue? Or do I need to have every technician , at the start of every shift, log into the support queue manually? As an aside , this community has been very helpful in getting my Asterisk box up and running. Thanks to all. Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple TDM400 vs T1
Another question: We have a main office with approx 10 incoming lines. Some of the lines are now in a rotary configuration. Does anyone have any advice on the Pros/Cons to moving to a T1 pipe and an appropriate Digium card? Will the T1 give me more flexibility with Asterisk vs POTS? TIA Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentCallBackLogin
I have the AgentCallBackLogin working well when the support technician logs into the queue manually. If there a way to get certain extensions to automatically log into the queue? That way I do not have to worry about help desk staff forgetting to log into the support queue and never receiving support calls. Thanks Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote Office question, Draytek , recommended analog phone
If there a way to connect an IP phone at a remote office to a device (like the Sipura 3000) that would allow local POTS failover in case of the Asterisk or VPN going down? I know the Sipura would allow a analog phone to connect to the local POTS in case of a * failure, we just want to standardize on IP phones ( Polycom 600) all around. Also has anyone had experience with the Draytek VOIP Wireless routers? And finally, if we need to use a Sipura 3000 in the remote offices is there any benefit with going with a Sayson analog phone versus any other? Thanks Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions.conf question
What is the best guide for the following situation? When a client dials a long distance # ( ie 5554441212) I want Asterisk to use my channel 4 on my TDM400 to dial. But , I want it to dial our local long distance exchange first. I have it dialing fine , I just need to know how to have it dial the exchange first . Something like 413,,5554441212. Thanks for the help Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-office topology suggestions
We are looking at putting Asterisk into use at our company. We have pushed it past proof of concept training and would like to roll it out in the very near future. One stumbling block remains: We have five offices in Canada. Our main office is in Edmonton , with branch offices all over the nation. I would like to place the Asterisk server in the Edmonton office and have it route calls to the branch offices. I would also like to have each of the branch offices have a local phone number. That local phone # would actually dial into the Asterisk box , and then routed appropriately via VPN to the correct location. This gives us a method of controlling and tracking all calls made to all offices. The issue is this: How can I have a phone number in a city over 1000 miles connect to the Asterisk box in an economical way? I have only tested the Asterisk box with a TDM11B and have no real experience with T1s . Would they help in this situation? Thanks in advance Shawn Dillon PS- My previous post on the issue of my TDM11B is now resolved. It was a dead FXO module. Thanks for the responses. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference
Thanks to all who have helped me build and test out Asterisk installation thus far. I needed to move my * installation to a new box , due to the fact my test machine would not support PCI 2.2 ( which I am told is required to use my TDM11B). I have * up and running and I am attempting to compile the app_conference source. The MeetMe app has too much echo. I am running Debian 2.4.26 and get tons of compile errors. If I compile right from the CVS of app_conference I get: chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: error: invalid option `abi=altivec' cc1: error: invalid option `dynamic-no-pic' cc1: error: unrecognized option `-faltivec' cc1: error: bad value (7450) for -mcpu= switch cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) make: *** [app_conference.o] Error 1 I then fix the mcpu ( I am on a Pentium4 Box). I comment out the line. I run make clean and make and then get the following. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [conference.o] Error 1 I change my two lines in conference.c as per http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html I run make clean, make and get the following error chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o member.o member.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) member.c: In function `member_exec': member.c:76: error: structure has no member named `dnid' member.c:76: error: structure has no member named `callerid' member.c:76: error: structure has no member named `ani' make: *** [member.o] Error 1 I have edited my member.c to remove any reference to the dnid,callerid and ani , and it compiles. But when someone connects to the conference * crashes. I know I am missing something simple ( I hope) I have also followed the instructions on http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html to no avail. I have also copied my app_conference files from another Asterisk box ( it compiles fine on that box). On my new box it will not compile. TIA Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk , ISA Firewall/VPN , STUN and other issues
I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone , on the SIP client nothing. The Debian machine has two NICs , one with a static external IP and one with an internal IP. Our remote offices are behind a mixture of firewalls. I have some questions with regards to our testing and setup. 1) Is there a way to get the SIP/IAX client to work via the VPN? This would be the easiest way. 2) If not can I install a STUN server on the same machine as the * server? Can it use the same internal and external IPs as the * server? 3) Is there a hardphone that supports VPN that has been tested? 4) What is the best hardphone to use with Asterisk? Thanks for the input Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
