[asterisk-users] PRI ANI/CallerID
For some reason something that seems like it should be simple is leaving me a bit perplexed. I am receiving incoming CallerID ANI on my PRI, but on my VoIP phones the display just shows asterisk when calls come in. I am receiving the calls with DNIS and have the DNIS digits setup as extensions. Do I need to add something to force relay the received caller ID to the phone? Any help is appreciated... Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Hardware
Hello I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. I appreciate your input. Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automon Filenames
Can someone give me some direction on automon filenames? I would like them to be the dialed number if possible. I saw a patch available for changing this but havent quite figured out how to use it. Can someone point me in the right direction? Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Non-PRI T1
We do have a PRI on order but have an immediate need so band-aid in the mean time. This circuit has actually been hooked to a channel bank for years with just analog phone and I stuck Asterisk in the middle about 3 months ago. I setup the standard S extension and it is handling incoming no problem. One other question - how do I get outgoing calls to select last available channel instead of first? Thanks, Dave David J. Sampson Information Technology Manager InnSeason Resorts 212 Mid-Tech Drive West Yarmouth, MA 02673 508-957-1881 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Fedyk Sent: Sunday, January 08, 2006 10:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Non-PRI T1 David Sampson wrote: Hello - I have a non-PRI T1 [...] How do I take incoming calls on these same channels? You should get a PRI T1. The minute you get close to capacity on this line you will run into timing issues with incoming and outgoing lines competing with each other. This problem will only happen when you need it most, which is the worst failure case. Unless your usage never spikes above 50% usage (counting incoming *and* outgoing) on this line you will regret not using a PRI. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Non-PRI T1
Hello I have a non-PRI T1 setup and have been making outgoing calls for several months no problem. I have Zapata.conf setup for fxs_ks on these channels. How do I take incoming calls on these same channels? Do I need to change the signaling? Any help is appreciated. Thank you, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
If I understand correctly you are supposed to patch the Makefile in the apps directory and then run the main Makefile. I've tried both ways - the patch failed on the main Makefile. Should I try to make that work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Friday, September 16, 2005 3:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem On Thu, 2005-09-15 at 16:10 -0400, David Sampson wrote: I’ve reduced my problem down to this: [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps# make Are you trying to use make from the apps directory? You have to run make from the main asterisk source directory. Look at the patch file necessary for the main Makefile. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
I used the latest version (.3) and also the previous .2 ver (pre20). The spandsp seems to compile but when I download the rxfax/txfax .c files and drop them in the apps directory that is where I get the compile error. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, September 14, 2005 2:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem What version of spandsp are you attempting to compile in to the 1.0.9 tree? On 9/14/05, David Sampson [EMAIL PROTECTED] wrote: Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
Ive reduced my problem down to this: [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps# make cc -D_GNU_SOURCE -o app_rxfax.so app_rxfax.c -lspandsp -ltiff app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make: *** [app_rxfax.so] Error 1 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Sampson Sent: Thursday, September 15, 2005 12:17 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem I used the latest version (.3) and also the previous .2 ver (pre20). The spandsp seems to compile but when I download the rxfax/txfax .c files and drop them in the apps directory that is where I get the compile error. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, September 14, 2005 2:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem What version of spandsp are you attempting to compile in to the 1.0.9 tree? On 9/14/05, David Sampson [EMAIL PROTECTED] wrote: Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] RxFax/TxFax - Compile Problem
Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hangup Faster
Hello My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesnt disconnect right away. Any ideas on how to resolve this? Thanks, Dave ___ 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] Channel Bank Help Please....
Hello I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone = us defaultzone=us span=1,1,0,esf,b8zs fxoks=1-24 zapata.conf [channels] group=1 language=en signalling=fxo_ks usecallerid=no context=default echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=1.0 txgain=1.0 channel = 1-24 extensions.conf exten = 3500,1,Dial,Zap/1|60 ; exten = 3500,2,Hangup exten = 3501,1,Dial,Zap/2|60 ; exten = 3501,2,Hangup exten = 3502,1,Dial,Zap/3|60 ; exten = 3502,2,Hangup exten = 3503,1,Dial,Zap/4|60 ; exten = 3503,2,Hangup exten = 3504,1,Dial,Zap/5|60 ; exten = 3504,2,Hangup exten = 3505,1,Dial,Zap/6|60 ; exten = 3505,2,Hangup exten = 3506,1,Dial,Zap/7|60 ; exten = 3506,2,Hangup exten = 3507,1,Dial,Zap/8|60 ; exten = 3507,2,Hangup exten = 3508,1,Dial,Zap/9|60 ; exten = 3508,2,Hangup exten = 3509,1,Dial,Zap/10|60 ; exten = 3509,2,Hangup When I attempt to call these extensions I get: *CLI dial 3501 -- Executing Dial(OSS/dsp, Zap/2|60) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 1: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 3: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 4: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 5: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 6: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 7: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 8: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 9: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 10: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 11: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 12: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 13: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 14: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 15: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 16: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 17: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 18: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 19: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 20: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 21: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 22: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 23: Yellow Alarm Aug 2 12:59:50 WARNING[3397]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 24: Yellow Alarm Aug 2 12:59:50 WARNING[3401]: chan_zap.c:3195 zt_handle_event: Detected alarm on channel 2: Yellow Alarm -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (local, 3501, 2) exited non-zero on 'OSS/dsp' Hangup on console Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 1 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 2 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 3 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 4 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 5 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 6 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 7 Aug 2 12:59:55 NOTICE[3397]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 8 Aug 2
[Asterisk-Users] Mitel SX2000 Integration
I have a Mitel SX2000 with no voicemail. Im wondering if it would be possible to use Asterisk to meet the need. This is a hotel property with mostly analog extensions. Dave ___ 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] Channel Banks
I have many old channel banks around that I would like to use to generate analog extensions. Will most channel banks work with Asterisk? Dave ___ 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] Incoming Calls
I have 2 4-port Digium FXS cards in my system. I would like to play a different recording based on which trunk rings. Any pointers? Thanks ___ 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] EM Tie Line
How do I setup my T1 card as an EM tie line? Any special configuration? ___ 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
RE: [Asterisk-Users] EM Tie Line
Title: Message I am connecting to an eON Millenium if anyone has done this. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Loretitsch Sent: Monday, May 23, 2005 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] EM Tie Line /etc/zapata.conf span=1,1,0,esf,b8zs (adjust to suit your t1 framing and encoding) em=1-24 /etc/asterisk/zaptel.conf [channels] echocancel=yes echotraining=400 context=default signalling=featd (this send *ani*dnis* to my PBX, otherwise use 'em_w' for this parameter callerid=asreceived musiconhold=default group=1 channel=1-24 Are you hooking up to a Definity by chance? I can send those configs too if you need em' Pretty easy actually. I couldn't do a PRI tie line because I'm out of processor channels on the Avaya side. Bummer! Good luck! -Matt -Original Message- From: David Sampson [mailto:[EMAIL PROTECTED] Sent: Monday, May 23, 2005 3:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] EM Tie Line How do I setup my T1 card as an EM tie line? Any special configuration? ___ 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] Public vs. Private Network
Hello I am looking at connecting 7 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for automatic failover. Should I remove one of my public connections and replace it with a private circuit for best quality? Thank you, Dave ___ 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
RE: [Asterisk-Users] Public vs. Private Network
Does anyone else have info regarding the port speed matching the CIR? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Thursday, May 19, 2005 11:55 AM To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Public vs. Private Network I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for automatic failover. Should I remove one of my public connections and replace it with a private circuit for best quality? To run VoIP over Frame Relay you need your Port Speed to be the same as your CIR. Cisco has extensive docs about this, but I'm too lazy to look them up right now. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 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] Call Recording
I run a call center and would like agents to be able to initiate call recording from a web page. Can anyone help? Thanks, Dave ___ 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] Immediate Answer
Hello . I currently have a generic modem card in my Asterisk system and will be upgrading to a Digium card next week. What I need to be able to do is immediate answer and play a greeting, then hang-up. Currently the system answer after two rings how can I minimize this? Thanks, Dave ___ 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] Immediate Answer
Hello My PSTN line answers after two rings and plays a greeting but I would like it to answer sooner. How can I make it answer immediately? Thanks, Dave David J. Sampson Information Technology Manager InnSeason Resorts 212 Mid-Tech Drive West Yarmouth, MA 02673 508-957-1881 ___ 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] IP500 Registration
Hello I have an IP500 (my first). The phone is up and running and I am able to make outgoing calls but I cant get the phone to register and take incoming calls. This is what my sip.conf looks like: [8503] type=user username=dave callerid=Dave Sampson 8503 secret=default host=dynamic dtmfmode=inband context=millenium mailbox=8503 defaultip=10.10.5.53 progressinband=no SIP debug shows: May 4 14:57:51 NOTICE[10797]: chan_sip.c:7691 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '10.10.5.53' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' Any help is greatly appreciated. No NAT here just on the private net. Thanks, Dave ___ 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] Incoming Not Answering
Hello I have just setup my first Asterisk box and Im having a great time. I am having a little trouble getting incoming calls to answer. This is what I see on the console: Apr 25 17:01:07 NOTICE[3514]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Apr 25 17:01:15 NOTICE[3515]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' I have the demo config files in place which show the s extension being answered and a message played but this is not happening. Any assistance is appreciated. Thank you, Dave ___ 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
