[asterisk-users] No zap* commands?
I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? ___ --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] Something is trashing /var/run
For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zaptel, wanpipe and asterisk on a fresh install of CentOS. I did the same thing on Friday and had the same problem, so I scrubbed the disk and tried again. Any ideas what's going on here ? Thanks, Jim. ___ --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] No zap* commands?
Conrad Wood wrote: On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modules.conf Said not found. So I checked, sure enough, it isn't anywhere to be found. So I looked in the zaptel source. Nothing there so I looked in the asterisk source. Ah! a chan_zap.c but no chan_zap.so. It doesn't seem to be compiled for some reason. There don't appear to be any comments in the README or INSTALL files about configuration parameters anywhere that have to be changed. So I trashed 1.2.12 asterisk and downloaded 1.2.13. And zaptel 1.2.10. I'll see what that does. Thanks, Jim. ___ --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] Something is trashing /var/run
Rodrigo Gonzalez wrote: mkdir /var/run/asterisk in /etc/asterisk/asterisk.conf change where you see /var/run with /var/run/asterisk Jim Lynch wrote: For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zaptel, wanpipe and asterisk on a fresh install of CentOS. I did the same thing on Friday and had the same problem, so I scrubbed the disk and tried again. Any ideas what's going on here ? Thanks, Jim. Ah! Thanks. That'll sure do it. Hope someone has filed a bug for that oversight. Jim. ___ --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] sip provider not working
I am getting a couple of messages in the log I don't understand. The first is: Unsupported transport 'UDP' The second is Oct 3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing... Oct 3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing... Oct 3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing... The second is repeated a number of times. I am unable to get any audio to or from my sip provider. I checked the firewall and the necessary ports are open. It used to work before I installed tribox. I guess I have to go back to [EMAIL PROTECTED] Thanks, Jim. ___ --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] Where are the kernel sources?
Matthew Thompson wrote: yum install kernel-devel Should do the trick. It did, thanks. Jim. ___ --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] Can't get second line of Sangoma A200 to work.
I set up extension 120 for the first and 121 for the second. The first
one works as expected but I can't get a dial tone on the second one. I
hear a buzzing in the second port much like the first, but no dial
tone. I have power since the dtmf keys work OK. I tried changing the
exten = 121,hint,ZAP/2 to exten = 121,hint,ZAP/1 not really knowing
what the number after the ZAP/ was, but that didn't work.
Can someone give me a clue as to the proper configuration for the second
port? I also can't get the fxo ports to work either but I'll wait to
ask that one later after I've played with them a bit. I'm using
Tribox. Another strange thing is that when I make a change and press
the red bar at the top of the page, then Asterisk completely quits
working until I do a /etc/init.d/asterisk stop and start. But I can
live with that.
Thanks,
Jim.
extensions_additional.conf
[ext-local]
include = ext-local-custom
exten = 101,1,Macro(exten-vm,101,101)
exten = 101,hint,SIP/101
exten = ${VM_PREFIX}101,1,Macro(vm,101,DIRECTDIAL)
exten = 102,1,Macro(exten-vm,novm,102)
exten = 102,hint,SIP/102
exten = 120,1,Macro(exten-vm,novm,120)
exten = 120,hint,ZAP/1
exten = 121,1,Macro(exten-vm,novm,121)
exten = 121,hint,ZAP/2
zapata_additional.conf
;;[120]
signalling=fxo_ks
record_out=Adhoc
record_in=Adhoc
[EMAIL PROTECTED]
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=120 120
busydetect=no
busycount=7
accountcode=
channel=1
;;[121]
signalling=fxo_ks
record_out=Adhoc
record_in=Adhoc
[EMAIL PROTECTED]
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=121 121
busydetect=no
busycount=7
accountcode=
channel=2
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[asterisk-users] Dial and connect to sip provider works, but no audio.
This is strange. I upgraded from an older [EMAIL PROTECTED] that was working to the latest Tribox. I also added a A204 board, but for some reason neither the Grandstream phone or a phone connected to the Linksys ATA has any audio either way via the Telasip connection. Audio works OK between the phones, so I'm pretty sure the extension configuration is OK.. Here's my sip configs. I added the [from-pstn] to this file because I didn't see it defined anywhere else. I realize it will go away when I change the extensions but it wasn't working so I thought I'd try it. I don't see much difference in configuration from when it worked and now, other than the missing [from-pstn] block. Thanks for any help. Jim. sip_additional.conf [EMAIL PROTECTED] [101] username=101 type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=101 101 [102] username=102 type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=102 102 [from-pstn] type=user qualify=yes insecure=very host=xxx.telasip.com [telasip] username=xxx type=friend secret=xxx.yyy qualify=yes insecure=very host=xxx.telasip.com fromuser=xxx fromdomain=xxx.telasip.com dtmfmode=rfc2833 context=from-pstn ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding nat=1 to each peer definition to ; solve translation problems. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying The number you have dialed is not in service. Please check the ; number and try again. context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf ~ ___ --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] How do I reset a password?
I've forgotten the user/pw for my freepbx adnim. I'm using [EMAIL PROTECTED] Is there a way to discover them or reset them. I have root access to the system. I did a google search but that didn't help. Thanks, Jim. ___ --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] How do I reset a password?
I'm looking for the username/password to access the web gui for freepbx admin rather than the voicemail passwords. I need to reconfigure the extentions/ring groups. Thanks, Jim. Doug Lytle wrote: Jim Lynch wrote: I've forgotten the user/pw for my freepbx adnim. I'm using [EMAIL PROTECTED] Is there a way to discover them or reset them. I have root access to the system. I did a google search but that didn't help. On a normal installation the passwords are located in the voicemail.conf file located in /etc/asterisk/voicemail.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Where are the kernel sources?
I need to build wanpipe to suport a Sangoma a200 card and when I do a Setup install as directed I get: Kernel source directory /lib/modules/2.6.9-42.0.2.EL/build not found I did a yum search kernel but didn't see anything that suggested the kernel source tree. Running the latest Tribox on centOS 4.4. I could use rpmfind, but I've not had a lot of luck ever getting an rpm that worked that way. Thanks, Jim. ___ --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] Sangoma A200 woes
It boots fine without the card. Jim. El Flynn wrote: Jim Lynch wrote: I attempted to install a new A200 module with one each FSX-2 and FXO-2 module. I connected an internal power connector to the board as instructed, but when the system reboots, it just beeps at me. It doesn't even let me get to a bios prompt. I removed both of the modules and it still behaves in the same way. The only other boards in the system are the ethernet and video boards. Does it still not boot when you've removed the card from the box? Flynn ___ --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] Sangoma A200 woes
I attempted to install a new A200 module with one each FSX-2 and FXO-2 module. I connected an internal power connector to the board as instructed, but when the system reboots, it just beeps at me. It doesn't even let me get to a bios prompt. I removed both of the modules and it still behaves in the same way. The only other boards in the system are the ethernet and video boards. The system is a Dell Dimension 440 system. Is there a requirement for a newer mother board to support that board? Any suggestions would be deeply appreciated. I deal with static sensitive parts on a regular basis and have both a special smock and an arm band so I don't think I've zapped anything. Thanks, Jim. ___ --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] trunk rollover
Jon Scottorn wrote:
What kind of line is being used?
in zapata.conf:
group = 1
channel = 1,3,5,6
I create a zap group will all your lines and dial out using the zap
group ie...
Dial(Zap/g1/${EXTEN})
By using the group it dials on the first available line.
If you want a more complex setup I have that as well.
I have an agi script that looks at the number dialed and determins if
it is a local call if so, dial out the ZAP line, if all ZAP lines are
busy dial out an IAX provider, I all IAX lines are busy, then roll to
my SIP provider.
Took a bit to figure it all out and get working but it is very useful.
Jon
Hi Jon,
Thanks, One of the lines is a sip connection to Telasip, the other is a
ZAP line. I'd appreciate any help I could get. I don't know what an
agi script is, so be gentle. :)
Thanks,
Jim.
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[Asterisk-Users] trunk rollover
I was hoping that rolling over to the next trunk would be simple, but it doesn't appear to be so, especially for a newbie. So I'm looking for a simple way where if I get a busy on the first outgoing trunk, I can do something to get connected to the next one. Perhaps something like the big boys do and dial 9 first? I'm guessing a custom dial plan might do that but I haven't figured out how to do it. I'm running [EMAIL PROTECTED] version 2.8 sounds right. (maybe that's asterisk 2.8) Can someone shed some light on a work around until I can figure out rollovers? Thanks, Jim. ___ --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] Can I get caller id passed to a phone connected to a Supura 2100?
I have a Uniden wireless phone connected into Linksys/Supura 2100. It works well, except I never see any caller ID information displayed on the phone. Is that a setting in the 2100 that I'm missing, or is it an Asterisk setting or isn't it possible? Thanks, Jim. ___ --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] Anyone with GSM488 experience?
I need another fxo line. Has anyone had any experience with connecting the gsm488 into asterisk? Thanks, Jim. ___ --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] Anyone with GSM488 experience?
Martin Joseph wrote: On Jun 8, 2006, at 6:49 AM, Jim Lynch wrote: I need another fxo line. Has anyone had any experience with connecting the gsm488 into asterisk? If you mean the Handytone 488 from grandstream, yes. It has some pretty major issues IMO. First of all, when the incoming calls through the FXO arrive, the FXS insists on ringing, and if you pick it up during or slightly after the first ring, the call is not answered and you are in no man's land. Thanks, I guess I'll do something different. Jim ___ --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] Two questions about [EMAIL PROTECTED] and backups.
Thanks to all who responded. Jim. Alex Robar wrote: Jim, There are SourceForge.net forums for [EMAIL PROTECTED] where you'll probably find better answers to your AAH questions. They are located here: https://sourceforge.net/forum/?group_id=123387 https://sourceforge.net/forum/?group_id=123387 In terms of backup, AAH has a built-in backup feature as part of the FreePBX GUI. Set your schedule, tell it which components (configs, voicemails, etc) you want to backup, and it'll generate a tarball that you can pull off the system at your leisure. Alex ___ --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] Two questions about [EMAIL PROTECTED] and backups.
First question, is there a forum for [EMAIL PROTECTED] specific questions? I've asked what must have been questions about [EMAIL PROTECTED] here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in /etc, where else does asterisk keep files that need to be backed up? Thanks, Jim. ___ --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] dnd error message in the log
Is this a problem? What is dnd anyway?Thanks,Jim.May 2 10:44:08 DEBUG[6277] db.c: Unable to find key 'SIP/201' in family 'dnd' ___ --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] Telasip config problem/question
I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air.I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.cominsecure=veryqualify=yestype=usercontext=from-pstnThen I went to incoming routes and set it up for any did, any cid, but the telasip connection is taking a different route. Here are the log entries for both. Telasip:May 2 11:11:55 DEBUG[2670] chan_sip.c: Checking SIP call limits for device jlynchMay 2 11:11:55 DEBUG[2670] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]May 2 11:11:55 VERBOSE[6343] logger.c: -- Executing Set(SIP/jlynch-cf63, FROM_DID=6782280738) in new stackMay 2 11:11:55 VERBOSE[6343] logger.c: -- Executing Answer(SIP/jlynch-cf63, ) in new stack May 2 11:11:55 VERBOSE[6343] logger.c: -- Executing PlayTones(SIP/jlynch-cf63, ring) in new stackMay 2 11:11:55 DEBUG[6343] channel.c: Scheduling timer at 160 sample intervalsMay 2 11:11:55 VERBOSE[6343] logger.c: -- Executing NVFaxDetect(SIP/jlynch-cf63, 0) in new stackMay 2 11:11:55 DEBUG[6343] app_nv_faxdetect.c: Preparing detect of fax (waitdur=4ms, sildur=1000ms, mindur=100ms, maxdur=-1ms) May 2 11:11:55 DEBUG[2670] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match FoundMay 2 11:11:55 DEBUG[2670] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED]' of Response 103: Match FoundMay 2 11:11:55 DEBUG[6343] channel.c: Scheduling timer at 0 sample intervals May 2 11:11:55 DEBUG[6343] app_nv_faxdetect.c: Got hangupZap:May 2 10:55:09 DEBUG[2670] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]May 2 10:55:09 VERBOSE[6287] logger.c: -- SIP/200-9b14 answered Zap/1-1May 2 10:55:09 DEBUG[6287] chan_zap.c: Requested indication -1 on channel Zap/1-1May 2 10:55:09 DEBUG[6287] channel.c: Scheduling timer at 0 sample intervalsMay 2 10:55:15 DEBUG[6287] channel.c: Didn't get a frame from channel: SIP/200-9b14May 2 10:55:15 DEBUG[6287] channel.c: Bridge stops bridging channels Zap/1-1 and SIP/200-9b14May 2 10:55:15 DEBUG[6287] chan_sip.c: update_call_counter(200) - decrement call limit counter May 2 10:55:15 DEBUG[6287] app_dial.c: Exiting with DIALSTATUS=ANSWER.May 2 10:55:15 VERBOSE[6287] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/1-1' in macro 'dial'May 2 10:55:15 VERBOSE[6287] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/1-1' in macro 'exten-vm'May 2 10:55:15 VERBOSE[6287] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/1-1'I even attempted to add an incoming route using the Telasip did, but it didn't work either. I have the radio button clicked in each of them to route the call to extension 200. ___ --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] Cant get voicemail
I've enabled voice mail for extension 200 in the extensions menu, and I've set the password to 1234. When I dial *97 which is listed as Your messages in the applications menu, it says Password I enter 1234 and it says, login incorrect, Password so what am I missing? Thanks,Jim. ___ --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] Cant get voicemail
OK, a combination of all the suggestions has pointed me in the right direction. The phone I'm using is the Budge Tone 100. For some reason it is not sending the dtmf tones. I had set the dtmfmode in the extensions menu to RFC2833, but I chanced to recall there was a dtmf mode in the phone config also. Sure enough, it was not set to RFC2833. When I changed that and rebooted the phone, it started working. Thanks to all. Does anyone else think rebooted my phone sounds a little funny? Thanks,Jim.On 5/1/06, Christian Buchter [EMAIL PROTECTED] wrote: Make sure *97 is indeed your voice mail number first. If you have remote voice-mail setup for users, you can just try it from outside(calling in). If that is indeed correct, then move another phone to that extension. At least you can narrow down thether it is the phone or the server causing the issue. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jim LynchSent: Monday, May 01, 2006 9:06 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Cant get voicemail I've enabled voice mail for extension 200 in the extensions menu, and I've set the password to 1234. When I dial *97 which is listed as Your messages in the applications menu, it says Password I enter 1234 and it says, login incorrect, Password so what am I missing? Thanks,Jim._This email has been scanned by MessageLabs on behalf of E-INS _ This email has been scanned by MessageLabs on behalf of E-INS ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Asterisk is stripping my area code
I've installed [EMAIL PROTECTED] and gotten inbound calls going to an extension, extension to extension calling works but I'm still missing a few pieces. The most annoying one is that apparently asterisk is stripping the area code from the number I'm dialing but I can't figure out how to stop it. I have in my outbound route under Dialing rules: 1NXXNXXNXXNXXWe are required to dial all 10 numbers since there are 3 area codes in Atlanta now.Using freePBX admin. I think [EMAIL PROTECTED] is version 2.7. One less than the most recent, in any case. Any suggestions would be helpful.Thanks,Jim. ___ --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] RE: Asterisk is stripping my area code
I don't know if this helps, from the log.Jim.Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Executing Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new stack Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Deferring dialing... Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 1/97707190239 Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, channel 1 Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) ___ --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] RE: Asterisk is stripping my area code
No, I'm just dialing 7707190239. When I tried it with a 1, it gave me the same result, a nice lady telling me when making a local call you must first dial the areacode or words to that effect. >From the log, after using the 1: Apr 30 13:29:04 DEBUG[4242] pbx.c: Function result is ' 7707190069'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing NoOp(SIP/200-fa0b, CallerID set to 7707190069) in new stack Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, GROUP()=OUT_1) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function result is '1'Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?108) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, DIAL_NUMBER=17707190239) in new stackApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, DIAL_TRUNK=1) in new stack Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing AGI(SIP/200-fa0b, fixlocalprefix) in new stackApr 30 13:29:04 VERBOSE[4242] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Apr 30 13:29:04 VERBOSE[4242] logger.c: fixlocalprefix: Removed prefix. New number: 7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c: -- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, OUTNUM=97707190239) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, custom=ZAP/1) in new stack Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?16) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c : Not taking any branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Dial(SIP/200-fa0b, ZAP/1/97707190239|120|W) in new stackApr 30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239' Apr 30 13:29:04 DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Called 1/97707190239Apr 30 13:29:05 DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05 DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel 1So what should I have in my dialing plan to let me dial 7707190239 and have it use that exact number? Or do I have to dial 9 first?Thanks, Jim.On 4/30/06, Kerry Garrison [EMAIL PROTECTED] wrote: Are you dialing 9 first? It is showing that the digits you dialed are: 9-770-719-0239 Using your dialplan you should be dialing 1-770-719-0239 Kerry GarrisonPublisher - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jim LynchSent: Sunday, April 30, 2006 10:10 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] RE: Asterisk is stripping my area code I don't know if this helps, from the log.Jim.Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Executing Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new stack Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Deferring dialing... Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, channel 1 Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] RE: Asterisk is stripping my area code
That was it! Thanks.Jim.On 4/30/06, Kerry Garrison [EMAIL PROTECTED] wrote: Do you have 9 as a prefix in the trunk? It is actually ADDING a 9 to the phone number before it dials. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jim LynchSent: Sunday, April 30, 2006 10:33 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] RE: Asterisk is stripping my area code No, I'm just dialing 7707190239. When I tried it with a 1, it gave me the same result, a nice lady telling me when making a local call you must first dial the areacode or words to that effect. From the log, after using the 1: Apr 30 13:29:04 DEBUG[4242] pbx.c: Function result is ' 7707190069'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing NoOp(SIP/200-fa0b, CallerID set to 7707190069) in new stack Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, GROUP()=OUT_1) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function result is '1'Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?108) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, DIAL_NUMBER=17707190239) in new stackApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, DIAL_TRUNK=1) in new stack Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing AGI(SIP/200-fa0b, fixlocalprefix) in new stackApr 30 13:29:04 VERBOSE[4242] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix Apr 30 13:29:04 VERBOSE[4242] logger.c: fixlocalprefix: Removed prefix. New number: 7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c: -- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, OUTNUM=97707190239) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, custom=ZAP/1) in new stack Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?16) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c : Not taking any branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Dial(SIP/200-fa0b, ZAP/1/97707190239|120|W) in new stackApr 30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239' Apr 30 13:29:04 DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Called 1/97707190239Apr 30 13:29:05 DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05 DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel 1So what should I have in my dialing plan to let me dial 7707190239 and have it use that exact number? Or do I have to dial 9 first?Thanks, Jim. On 4/30/06, Kerry Garrison [EMAIL PROTECTED] wrote: Are you dialing 9 first? It is showing that the digits you dialed are: 9-770-719-0239 Using your dialplan you should be dialing 1-770-719-0239 Kerry GarrisonPublisher - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Jim LynchSent: Sunday, April 30, 2006 10:10 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] RE: Asterisk is stripping my area code I don't know if this helps, from the log.Jim.Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Executing Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new stack Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Deferring dialing... Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, channel 1 Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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
