Re: [Asterisk-Users] [OT] Looking for Web based SIP endpoint
Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. We tried to get it working with two developers without luck... I think it still has some way to go. Any reason you are looking for SIP and not IAX? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
Hi Matt: That suggestion is possibly on the right track. It made me remember that - although I'm not using Fedora, but SuSE 9.3, that it went through an automatic network update just recently. After that, I tried updating the Zaptel files from CVS and recompiling everything, but to no avail. The same error still occured. I eliminated hardware by swapping out a working TDM400 with the same FXS/FXO configuration. The same error occurs. The SuSE update may have moved some of the required files, although there are no complaints during the build and I can't determine what may have moved. I have still present and installed Bison 1.875, OpenSSL and zlib-devel, and of course Linux source for this SuSE disto. I'm completely faklempt! Can someone shed light on this delima?? Thanks so much if you can. I want my As-terisk back!!! It was working, damnit. Thanks, Matt for your suggestion. Scott At 09:30 PM 8/19/2005, you wrote: [EMAIL PROTECTED] wrote: Hi: I hope that someone can help with this problem that came up suddenly. I Did you upgrade Fedora Core? Check if the udev files still contain the required entries (normally fedora copies the old ones to 50-udev-rules.old and makes new ones). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 Scott Brown CTO Anderson Interactive Systems, Inc. 831-479-4493 office 831-419-4733 cell 831-479-4447 fax [EMAIL PROTECTED] ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
On Saturday 20 August 2005 09:58, Scott Brown wrote: Hi Matt: That suggestion is possibly on the right track. It made me remember that - although I'm not using Fedora, but SuSE 9.3, that it went through an automatic network update just recently. After that, I tried updating the Zaptel files from CVS and recompiling everything, but to no avail. The same error still occured. I eliminated hardware by swapping out a working TDM400 with the same FXS/FXO configuration. The same error occurs. The SuSE update may have moved some of the required files, although there are no complaints during the build and I can't determine what may have moved. Currently I am doing the following on SuSE : First reboot the PC with asterisk disabled. This will force the creation of the devices during boot from the /etc/udev/rules.d files. Try modprobing : modprobe -v -n wctdm This does nothing but tells you what would happen. If your modprobe.d/zaptel file is correct the the output from this command will be loading of zaptel,wcfxs and an execution of ztcfg. In other words you do not have to modprobe more than one module - dependencies are sorted by the modprobe.d/zaptel file. If you want -vv on the ztcfg file edit modprobe.d/zaptel. I remember from the wiki somewhere that one must not execute ztcfg more than once and this will happen if you modprobe zaptel and then wctdm and then execute ztcfg manually. So to load modprobe -v wctdm and to unload modprobe -v -r wctdm Add these commands to your asterisk startup script This does not seem to work on Fedora - SuSE has a section on bootup 'Creating devices' that seems to pre-initialise everything - Fedora seems not to have this so one has to resort to insmodding and sleeping viz: insmod zaptel sleep 3 insmod wcfxs sleep 3 ztcfg -vv The above has been learned from experience - the loading of the driver modules for asterisk/zap seems fraught with reliability issues - in some cases the drivers load without error - at other times various errors occur - I have not yet figured out the cause of failures Paul -- Paul Hewlett (Linux #359543) Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ 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] CALL QUALITY PROBLEM...
Hi Some basic mailing lists ethics: 1. writing in CAPITAL LETTERS usually indicates SHOUTING. Please don't do that. 2. when you want to start a new message to the list, write a new message, and don't just reply to an existing list message. 3. Proper English is also preffered, so readers spend less time on trying to understand your English and more on trying to help you. (/me no native English speaker and I know it shows well on my messages. I try, though). For the convinience of the readers, quoted message was converted to small caps (gu). See reply below, On Fri, Aug 19, 2005 at 07:18:44PM -0700, Ing. Marlo R. Beltran G wrote: hi i just implemented asterisk and is such a grate solution...i am using it polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port linksys switch... the problem is when somebody inside the network is making a call to other extension (in the same network) and for example is sending an e mail to the internet the quality goes down...it hears rally bad... i am on a 10/100 network with cat5e on cable, and switches...what can i do to have an excelent voice quality inside my network??? Do you actually use a 100Mbit full-duplex network? mii-tool is the simplest way to check that on Linux. If your card does not support it, maybe the messages in dmesg will tell. Also: does the relevant mail go through the computer that runs Asterisk? If so, all the switching may be irrelevant. In that case this may also be due to CPU usage issues rather than a network issue. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Quality problem on LAN when using the network!
Hi i just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail trough internet the quality goes down...it hears rally bad... i am on a 10/100 network with cat5e on wire, and switches...what can i do to have an excelent voice quality inside my network??? The e mail doesnt go trough the asterisk computer. Marlo ___ 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 quality problem when using lan
Hi i just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail trough internet the quality goes down...it hears rally bad... i am on a 10/100 network with cat5e on wire, and switches...what can i do to have an excelent voice quality inside my network??? The e mail doesnt go trough the asterisk computer. Marlo ___ 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] Call quality problem when using lan
On 03:57, Sat 20 Aug 05, Ing. Marlo R. Beltran G wrote: Hi i just implemented asterisk and is such a grate solution...i am using polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port linksys switch... the problem come when somebody inside the network is making a call to other extension (in the same network) and is sending an e mail trough internet the quality goes down...it hears rally bad... i am on a 10/100 network with cat5e on wire, and switches...what can i do to have an excelent voice quality inside my network??? The e mail doesn't go trough the asterisk computer. Hi, Maybe QoS is the answer to this problem. We have a lot more traffic on our lan, and we made all terminals QoS aware. We are giving the voice packets priority and even a ping -f cannot influence call quality. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Realtime sip_buddies register= how?
Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use register=.. on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions ? thanks Billy ___ 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] [OT] Looking for Web based SIP endpoint
Matt Riddell [EMAIL PROTECTED] wrote: Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Any reason you are looking for SIP and not IAX? Is there an IAX alternative that you'd recommend? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
Seems the latest distro's have changed the layout of the linux source tree needed to compile the zaptel stuff. I'm using FC3, upgraded the kernel, and have the same issue. Was able to install new sources, but they too are completely different tree layout compared to earlier stuff. The same is apparently happening with other distro's as well. There has been a bug item open for last several weeks relative to reworking the make files for these items. That suggestion is possibly on the right track. It made me remember that - although I'm not using Fedora, but SuSE 9.3, that it went through an automatic network update just recently. After that, I tried updating the Zaptel files from CVS and recompiling everything, but to no avail. The same error still occured. I eliminated hardware by swapping out a working TDM400 with the same FXS/FXO configuration. The same error occurs. The SuSE update may have moved some of the required files, although there are no complaints during the build and I can't determine what may have moved. I have still present and installed Bison 1.875, OpenSSL and zlib-devel, and of course Linux source for this SuSE disto. I'm completely faklempt! Can someone shed light on this delima?? Thanks so much if you can. I want my As-terisk back!!! It was working, damnit. Thanks, Matt for your suggestion. Scott At 09:30 PM 8/19/2005, you wrote: [EMAIL PROTECTED] wrote: Hi: I hope that someone can help with this problem that came up suddenly. I Did you upgrade Fedora Core? Check if the udev files still contain the required entries (normally fedora copies the old ones to 50-udev-rules.old and makes new ones). ___ 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] Asterisk Zaptel Leading Zero Problem With TE110P
Hi All I am having another strnage problem :) When I dialout on any number from asterisk, it use to add a leading zero in dialed number for e.g I dial a number 5832876 and when I check the tracer's result of PSTN switch that shows me call request for 05832876 thats why I can dial NWD and ISD calls but unable to dial local numbers Thanks Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] [OT] Looking for Web based SIP endpoint
Kevin Walsh wrote: Matt Riddell [EMAIL PROTECTED] wrote: Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Any reason you are looking for SIP and not IAX? Is there an IAX alternative that you'd recommend? Mozilla/FireFox:(PC/Linux) http://moziax.mozdev.org/ Java (Mac/PC/Linux):http://www.hem.za.org/jiaxclient/ ActiveX: (PC - IE) http://www.geocities.com/babarnazmi/index2.htm Think that should just about cover your bases. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Asterisk transcoding /Routing
Hello, Asterisk is said to handle call routing and codec translation. I would like to force transcoding function with asterisk but when I try to force transcoding I get the errors: codec not compatible or WARNING[4425]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/xxx compatible with SIP/yyy How exactly works asterisk, in order to transcoding? If you have any suggestions, hints, work around tricks I would appreciate them much Thanks in advance. George ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.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] What is the reason for warning Unable to allocate socket
hello i m getting follwing messages in asterisk-1.0.9 what is the reason can u pls tel me how to solve this Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new: Unable to allocate socket: Too many open files Aug 20 13:06:09 WARNING[7706]: channel.c:311 ast_channel_alloc: Alert pipe creation failed! Aug 20 13:06:09 WARNING[7706]: chan_sip.c:2081 sip_new: Unable to allocate channel structure Aug 20 13:06:09 NOTICE[7706]: chan_sip.c:7469 handle_request: Unable to create/find channel Aug 20 13:06:22 WARNING[7706]: acl.c:216 ast_lookup_iface: Unable to get IP of eth0: Bad file descriptor A __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] static noise with TDM revision G but not with revision F
Hi We have asterisk installed on a system and we had static noise We changed the TDM card from a revision G to a revision F and the static noise is gone Any idea what is the difference between the two revisions that could make this problem ? Is there a way to downgrade the cards that we have bought Patrick ___ 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] ViaTalk Down?
Is anyone else with ViaTalk experiencing an outage right now? My DID has been down since 5AM (8/20). Asterisk is unable to re-register or connect for outbound calls. I have also tried calling support and their number gives a fast busy. ___ 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] ViaTalk Down?
I'm a ViaTalk system engineer. I just got up, I'm about to check it out. Thanks for the heads up, I wouldn't have seen this until later. I can tell you however, that our monitoring system did not kick any messages to me about it acting funny in any way. I'll check it out and get back to you. Sherwood McGowan ViaTalk --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Justin Richards -Sent: Saturday, August 20, 2005 8:29 AM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] ViaTalk Down? - -Is anyone else with ViaTalk experiencing an outage right now? - My DID has been down since 5AM (8/20). Asterisk is unable to -re-register or connect for outbound calls. I have also tried -calling support and their number gives a fast busy. -___ -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
RE: [Asterisk-Users] ViaTalk Down?
I've restarted our switch via restart command from the CLI. Anyone have a quickie answer as to why asterisk would suddenly just stop responding? I was able to issue the restart command but I couldn't do sip show peer num and couldn't show channels, etc This is very disconcerting We've overall had little to now major issues with it running on our switch --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Justin Richards -Sent: Saturday, August 20, 2005 8:29 AM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] ViaTalk Down? - -Is anyone else with ViaTalk experiencing an outage right now? - My DID has been down since 5AM (8/20). Asterisk is unable to -re-register or connect for outbound calls. I have also tried -calling support and their number gives a fast busy. -___ -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
RE: [Asterisk-Users] ViaTalk Down?
Last note on this, I figured out it was due a freeze in registrations that we've been having an issue with on asterisk. I'm writing a custom monitoring script using sipsak for testing registrations, which would SMS the engineering dept when registrations stop working. Cheers, Sherwood McGowan --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Justin Richards -Sent: Saturday, August 20, 2005 8:29 AM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] ViaTalk Down? - -Is anyone else with ViaTalk experiencing an outage right now? - My DID has been down since 5AM (8/20). Asterisk is unable to -re-register or connect for outbound calls. I have also tried -calling support and their number gives a fast busy. -___ -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
Re: [Asterisk-Users] ViaTalk Down?
Thanks for the quick response (and call), its running again! On 8/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote: I've restarted our switch via restart command from the CLI. Anyone have a quickie answer as to why asterisk would suddenly just stop responding? I was able to issue the restart command but I couldn't do sip show peer num and couldn't show channels, etc This is very disconcerting We've overall had little to now major issues with it running on our switch --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Justin Richards -Sent: Saturday, August 20, 2005 8:29 AM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] ViaTalk Down? - -Is anyone else with ViaTalk experiencing an outage right now? - My DID has been down since 5AM (8/20). Asterisk is unable to -re-register or connect for outbound calls. I have also tried -calling support and their number gives a fast busy. -___ -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 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] Where did my DID's go??
Your only recourse is to get your new carrier to realize that the numbers have been released and to proceed with the porting despite the fact that they have not received the notification. Thanks for the info! I've forwarded your message to the new carrier in hopes that they'll be able to do something. Hatton ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
concur that the best way around this is to perioically restart. FWIW this is my restart script which I invoke from cron in the middle of the night... #!/bin/bash ASTERISK=/usr/sbin/asterisk RMMOD=/sbin/rmmod MODPROBE=/sbin/modprobe ZTCFG=/sbin/ztcfg echo Stopping $ASTERISK -rx stop when convenient if [ $? == 0 ]; then #let COUNT=0 while true; do echo waiting for asterisk to stop if [ ! -e /var/run/asterisk.pid ]; then break; fi sleep 5; echo . # let COUNT=COUNT+1 # if [ $COUNT == 40 ]; then # break; # fi done fi echo unloading modules $RMMOD wctdm sleep 5 $RMMOD zaptel sleep 15 killall -9 mpg123 echo loading modules $MODPROBE zaptel sleep 15 $MODPROBE wctdm sleep 15 $ZTCFG -v sleep 5 echo Starting $ASTERISK mark On 8/20/05, Rich Adamson [EMAIL PROTECTED] wrote: Seems the latest distro's have changed the layout of the linux source tree needed to compile the zaptel stuff. I'm using FC3, upgraded the kernel, and have the same issue. Was able to install new sources, but they too are completely different tree layout compared to earlier stuff. The same is apparently happening with other distro's as well. There has been a bug item open for last several weeks relative to reworking the make files for these items. That suggestion is possibly on the right track. It made me remember that - although I'm not using Fedora, but SuSE 9.3, that it went through an automatic network update just recently. After that, I tried updating the Zaptel files from CVS and recompiling everything, but to no avail. The same error still occured. I eliminated hardware by swapping out a working TDM400 with the same FXS/FXO configuration. The same error occurs. The SuSE update may have moved some of the required files, although there are no complaints during the build and I can't determine what may have moved. I have still present and installed Bison 1.875, OpenSSL and zlib-devel, and of course Linux source for this SuSE disto. I'm completely faklempt! Can someone shed light on this delima?? Thanks so much if you can. I want my As-terisk back!!! It was working, damnit. Thanks, Matt for your suggestion. Scott At 09:30 PM 8/19/2005, you wrote: [EMAIL PROTECTED] wrote: Hi: I hope that someone can help with this problem that came up suddenly. I Did you upgrade Fedora Core? Check if the udev files still contain the required entries (normally fedora copies the old ones to 50-udev-rules.old and makes new ones). ___ 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 -- regards, Mark P. Edwards FWD: 667917 ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
On Sun, Aug 21, 2005 at 12:32:06AM +1000, Mark Edwards wrote: concur that the best way around this is to perioically restart. This is ignoring the problem rather than solving it. If you both rmmod zaptel and restart asterisk, why not simply reboot? All of those sleeps there produce a nice downtime. See remarks below. FWIW this is my restart script which I invoke from cron in the middle of the night... #!/bin/bash ASTERISK=/usr/sbin/asterisk RMMOD=/sbin/rmmod MODPROBE=/sbin/modprobe ZTCFG=/sbin/ztcfg echo Stopping $ASTERISK -rx stop when convenient if [ $? == 0 ]; then #let COUNT=0 why do math in shell scripts? for i in `seq 40` while true; do echo waiting for asterisk to stop if [ ! -e /var/run/asterisk.pid ]; then break; fi A sudden 'killall -9 asterisk' by someone and that cron job will be in an endless loop. sleep 5; echo . # let COUNT=COUNT+1 # if [ $COUNT == 40 ]; then # break; # fi done fi echo unloading modules $RMMOD wctdm sleep 5 $RMMOD zaptel sleep 15 killall -9 mpg123 echo loading modules $MODPROBE zaptel sleep 15 $MODPROBE wctdm sleep 15 $ZTCFG -v sleep 5 echo Starting $ASTERISK -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] OT? ... Trying to get cid_rewrite script to work
Sorry if this is a resend, but it didn't appear to go the first time. Sorry if this is not the correct place to post this. I have downloaded the cid_rewrite scripts that are located at: http://www.muware.com/asterisk/ to my AAH v1.1 system. I apologize for my ignorance, but it says that I need to modify the agi_config.php, but doesn't indicate what I need to modify it to look like. (I'm not a dba kind of person, so this is a bit confusing) Has anyone had any luck getting these scripts working on AAH ? I do miss my call id with name that the telco used to provide. Thanks in advance, Alan ___ 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] ISDN BRI voice one way only
hi PSTN -- [Teles ISDN / Asterisk] -- SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Setup: * Teles 16.3 ISA ISDN card with hisax kernel module * Asterisk 1.0.9 (debian unstable package) with isdn4linux * Kernel 2.6.12 * P2 - 333 Mhz / 512 Mb RAM I found that some persons have had the same problem but no hint for solution. Does anybody know how to fix this? rgrds, Klem ___ 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] Searching For a Voip Provider
Why? --- Innocent Evil [EMAIL PROTECTED] wrote: Please change the subject to 'Advertisement of a VoIP Provider' -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 18 Aug 2005 11:55:50 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Searching For a Voip Provider Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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 __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] ATA186 reguest problem
hi, my ATA186 confige as SIP(600) on my Asterisk ,it only can be called in , but can not call out . between ATA186 and astersik there is aVPNon two netscreen 5gt. who can showme some idea ? ATA 186 configure same as SIP.conf SIP.conf on Asterisk : [general]port = 5060 ; Port to bind tobindaddr = 0.0.0.0 ; Address to bind todisallow=allallow=ulawcontext = local ; Default for local calls [600]type=friendusername=600secret=mondayhost=dynamicdefaultip=192.168.33.100canreinvite=no ; Cisco poops on reinvite sometimesqualify=600 ; Qualify peer is no more than 200ms awaydtmfmode=rfc2833callerid = SZ 600callgroup = 10pickupgroup = 10mailbox=600 [601]type=friendusername=601secret=mondayhost=dynamicdefaultip=192.168.33.100canreinvite=no ; Cisco poops on reinvite sometimesqualify=600 ; Qualify peer is no more than 200ms awaydtmfmode=rfc2833callerid = SZ 601callgroup = 10pickupgroup = 10mailbox=601 ON SIP debug mode shows: to 192.168.33.100:5060Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phoneCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEContact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udpUser-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)Expires: 300Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTERContent-Length: 274Content-Type: application/sdp v=0o=600 50100 50100 IN IP4 192.168.33.100s=ATA186 Callc=IN IP4 192.168.33.100t=0 0m=audio 1 RTP/AVP 0 18 8 101a=rtpmap:0 PCMU/8000/1a=rtpmap:18 G729/8000/1a=fmtp:18 annexb=yesa=rtpmap:8 PCMA/8000/1a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15 12 headers, 12 linesIgnoring this requestReliably Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phone;tag=as74d2a1cbCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXContact: Proxy-Authenticate: Digest realm="asterisk", nonce="1220cba1"Content-Length: 0 to 192.168.33.100:5060Retransmitting #1 (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phone;tag=as74d2a1cbCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXContact: Proxy-Authenticate: Digest realm="asterisk", nonce="7e0a728d"Content-Length: 0 Leng9to 192.168.33.100:5060Retransmitting #2 (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phone;tag=as74d2a1cbCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXContact: Proxy-Authenticate: Digest realm="asterisk", nonce="0d6babf9"Content-Length: 0 to 192.168.33.100:5060Retransmitting #1 (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phone;tag=as74d2a1cbCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXContact: Proxy-Authenticate: Digest realm="asterisk", nonce="1220cba1"Content-Length: 0 IP/2to 192.168.33.100:5060Retransmitting #2 (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phone;tag=as74d2a1cbCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXContact: Proxy-Authenticate: Digest realm="asterisk", nonce="7e0a728d"Content-Length: 0 Leng9to 192.168.33.100:5060Retransmitting #3 (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.33.100:5060From: sip:[EMAIL PROTECTED];user=phone;tag=2459813530To: sip:[EMAIL PROTECTED];user=phone;tag=as74d2a1cbCall-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXContact: Proxy-Authenticate: Digest realm="asterisk", nonce="0d6babf9"Content-Length: 0 to 192.168.33.100:5060Sip read: REGISTER sip:192.168.1.50 SIP/2.0Via: SIP/2.0/UDP 192.168.1.58:5060From: sip:[EMAIL PROTECTED];user=phone;tag=1707128448To: sip:[EMAIL PROTECTED];user=phoneCall-ID: [EMAIL PROTECTED]CSeq: 273 REGISTERContact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;expires=120User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)Content-Length: 0 ___ 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] Realtime sip_buddies register= how?
You can store your entire sip.conf using RealTime. That should allow for register = to work. -Matthew From: Guillermo Krepper [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 20 Aug 2005 13:05:02 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Realtime sip_buddies register= how? Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use register=.. on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions ? thanks Billy ___ 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] ZAP divert problem
I have a TDM400 running telco lines on ZAP2-4 My after hours config is supposed to receive the incoming call then divert it to my home phone by calling out one of the other zap channels available. console output as such... Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, ZAP/G1/0823274210) in new stack Aug 20 19:06:44 NOTICE[646]: app_dial.c:777 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time Aug 20 19:06:55 WARNING[646]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' in context 'incoming-pstn' -- Hungup 'Zap/3-1' extensions.conf. [incoming-pstn] include = outgoing-cell include = open|08:00-16:59|mon-fri|*|* include = open|08:00-12:29|sat|*|* include = closed|17:00-7:59|mon-fri|*|* include = closed|12:30-7:59|sat|*|* include = closed|*|sun|*|* [open] works fine... [closed] exten = s,1,Dial(ZAP/G1/0823274000) zapata.conf... [channels] signalling=fxs_ks context=incoming-pstn group=1 callgroup=1 pickupgroup=1 usecallerid=no faxdetect=incoming callerid=Incoming call on 2133878 rxgain=6.0 txgain=6.0 channel = 2 signalling=fxs_ks context=incoming-pstn group=1 callgroup=1 pickupgroup=1 usecallerid=no faxdetect=incoming callerid=Incoming call on 2133478 rxgain=6.0 txgain=6.0 channel = 3 signalling=fxs_ks context=incoming-pstn group=1 callgroup=1 pickupgroup=1 usecallerid=no faxdetect=incoming callerid=Incoming call on 2133479 rxgain=6.0 txgain=6.0 channel = 4 Anyone know what i am doing wrong? Leon -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 15/08/2005 ___ 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] Ring more than two isdn phones simultaneously
I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the first two specified phones ring. Etc. with the first channel 21 ist called, with the second 22. How can I get asterisk to signal to all phones with just one isdn channel? I am trying to duplicate the setup I had with my old isdn pbx with did above trick just fine... Maybe somebody can help me configure asterisk appropriately? Cheers, Arik PS: I gave following a try but without success: Dial(Zap/g1/21-29,,t) Dial(Zap/g1/21+29,,t) ___ 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] [OT] Looking for Web based SIP endpoint
The java client you mention states on it's webpage it has to install a local .dll/.so and that it only works for x86 Windows or Linux. Does anyone know of one that's completely in Java? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Saturday, August 20, 2005 4:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] Looking for Web based SIP endpoint Kevin Walsh wrote: Matt Riddell [EMAIL PROTECTED] wrote: Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Any reason you are looking for SIP and not IAX? Is there an IAX alternative that you'd recommend? Mozilla/FireFox:(PC/Linux) http://moziax.mozdev.org/ Java (Mac/PC/Linux):http://www.hem.za.org/jiaxclient/ ActiveX: (PC - IE) http://www.geocities.com/babarnazmi/index2.htm Think that should just about cover your bases. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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
Re: [Asterisk-Users] Ring more than two isdn phones simultaneously
how many connection do you have from your asterisk to the old pbx? i think on 1 ISDN connection its only possible to let 2 phones ring, because 1 ISDN 2 channels... Nico - Original Message - From: Arik Funke [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, August 20, 2005 7:44 PM Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the first two specified phones ring. Etc. with the first channel 21 ist called, with the second 22. How can I get asterisk to signal to all phones with just one isdn channel? I am trying to duplicate the setup I had with my old isdn pbx with did above trick just fine... Maybe somebody can help me configure asterisk appropriately? Cheers, Arik PS: I gave following a try but without success: Dial(Zap/g1/21-29,,t) Dial(Zap/g1/21+29,,t) ___ 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
Re: [Asterisk-Users] Echo cancellation again ...
Aug 17, 2005, at 5:44 AM, Tom Hayden wrote: I have experienced pretty nasty echo on my PRI w/TE110P. The echo was only coming from other POTS lines, because cell phones already have echo cancellation, and other PBX's had the same. I resolved the problem by turning on the AGGRESSIVE option and it works fine now, and we haven't noticed a severe degradation in sound quality - most of my operators were just happy the echo was gone :) +1 here too: Uncommenting AGGRESSIVE_SUPPRESSOR and recompiling took care of 99% of my TE110P/PRI echo. -Rob. -- Robert Goodyear Brand Up LLC http://www.brand-up.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] Sudenly unable to get incoming from Broadvoice
I think I spotted what's going on. When I sift through the sip debug I see that my server is looking for my number in the context I take the calls into. Problem is its not there. I was expecting * to dump the call into the exten=s,1,blahblahblah logic but it's not doing that. Now I think about it I did a cvs code update/co,pile etc about 5 days ago. Have I discovered a newly intor'd bug? I'm gonna roll back and see if it goes away. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Lock Extension
On Aug 18, 2005, at 3:07 AM, Stephen wrote: Hi All, How can I lock the extension in Asterisk? For example , my extension is 1000 and I am away for business trip. I want to lock my extension during my absence. Can it be done in Asterisk? regards, Stephen You could write a little script to mangle/unmangle your SIP context and then SIP RELOAD. You could assign it to a context called 'disabled' whose only valid extension matching therein is to that same macro to authenticate and change your context back. ___ 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] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten = *97,1,Answer exten = *97,2,VoicemailMain([EMAIL PROTECTED]) exten = *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing Answer(SIP/200-d83a, ) in new stack Aug 20 18:57:45 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-d83a' -- Executing Answer(SIP/200-81f6, ) in new stack Aug 20 18:57:59 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-81f6' -- Executing Answer(SIP/201-a86c, ) in new stack Aug 20 19:00:24 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/201-a86c' -- Executing Dial(SIP/201-1e08, SIP/200|20|Ttm) in new stack -- Called 200 -- Started music on hold, class 'default', on SIP/201-1e08 -- SIP/200-b925 is ringing -- Stopped music on hold on SIP/201-1e08 -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/201-1e08, su200) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav49, 0x818eb40 -- x=1, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: gsm, 0x813a7e8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav, 0x818ed88 -- User hung up == Spawn extension (default, 200, 2) exited non-zero on 'SIP/201-1e08' -- Executing Answer(SIP/200-4b1a, ) in new stack Aug 20 19:01:57 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-4b1a' -- Executing Answer(SIP/200-5369, ) in new stack Aug 20 19:02:11 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-5369' linux*CLI Angus ___ 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] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
You spelled Voicemailmain wrong somewhere. Or your extensions are not in sync with the conf file. Verify that the extensions.conf is correct then 'extensions reload'. You can also do show dialplan context to view what is currently loaded in memory. -Matthew From: Angus Comber [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 20 Aug 2005 19:21:07 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten = *97,1,Answer exten = *97,2,VoicemailMain([EMAIL PROTECTED]) exten = *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing Answer(SIP/200-d83a, ) in new stack Aug 20 18:57:45 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-d83a' -- Executing Answer(SIP/200-81f6, ) in new stack Aug 20 18:57:59 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-81f6' -- Executing Answer(SIP/201-a86c, ) in new stack Aug 20 19:00:24 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/201-a86c' -- Executing Dial(SIP/201-1e08, SIP/200|20|Ttm) in new stack -- Called 200 -- Started music on hold, class 'default', on SIP/201-1e08 -- SIP/200-b925 is ringing -- Stopped music on hold on SIP/201-1e08 -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/201-1e08, su200) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav49, 0x818eb40 -- x=1, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: gsm, 0x813a7e8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav, 0x818ed88 -- User hung up == Spawn extension (default, 200, 2) exited non-zero on 'SIP/201-1e08' -- Executing Answer(SIP/200-4b1a, ) in new stack Aug 20 19:01:57 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-4b1a' -- Executing Answer(SIP/200-5369, ) in new stack Aug 20 19:02:11 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-5369' linux*CLI Angus ___ 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
Re: {Scanned} RE: [Asterisk-Users] [OT] Looking for Web based SIP endpoint
Pulu Anau wrote: The java client you mention states on it's webpage it has to install a local .dll/.so and that it only works for x86 Windows or Linux. Does anyone know of one that's completely in Java? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Saturday, August 20, 2005 4:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] Looking for Web based SIP endpoint Kevin Walsh wrote: Matt Riddell [EMAIL PROTECTED] wrote: Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Any reason you are looking for SIP and not IAX? Is there an IAX alternative that you'd recommend? Mozilla/FireFox:(PC/Linux) http://moziax.mozdev.org/ Java (Mac/PC/Linux):http://www.hem.za.org/jiaxclient/ ActiveX: (PC - IE) http://www.geocities.com/babarnazmi/index2.htm Think that should just about cover your bases. :) Try this site http://www.microappliances.com the only problem is that it uses IE(active x) Tom -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Thank You For Choosing Cache Communications ___ 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] ViaTalk Down?
Sherwood McGowan wrote: Anyone have a quickie answer as to why asterisk would suddenly just stop responding? I was able to issue the restart command but I couldn't do sip show peer num and couldn't show channels, etc This is very disconcerting Your SIP channel driver was deadlocked. This can happen for a number of reasons, but all of them are bad, and need to be fixed. Depending on the Asterisk version you are running, there may be some known situations under which they can occur; as best we can tell, there are none left in CVS HEAD related to chan_sip. ___ 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] X100P compatible
Why my X100P detect the ring after 3 o 4 seconds? The funny thing that when I have an incoming call asterisk receive a signal but the commands start after 3 or 4 seconds. Moreover, when the call end the hungup has the same delay. any ideas? ___ 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] Asterisk Zaptel Leading Zero Problem With TE110P
On Sat, 20 Aug 2005, Gulzar Hussain wrote: I am having another strnage problem :) When I dialout on any number from asterisk, it use to add a leading zero in dialed number for e.g I dial a number 5832876 and when I check the tracer's result of PSTN switch that shows me call request for 05832876 thats why I can dial NWD and ISD calls but unable to dial local numbers What channel do you use? For chan_zap you may want to look at the pridialplan, especially pridialplan=dynamic and the nationalprefix etc. Peter ___ 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] Ring more than two isdn phones simultaneously
On Sat, 20 Aug 2005, Nico Giefing wrote: how many connection do you have from your asterisk to the old pbx? i think on 1 ISDN connection its only possible to let 2 phones ring, because 1 ISDN 2 channels... This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel for each destination at the time of the CONNECT message. In the isdn world it is common to not actually allocate a B-channel until it is needed to carry audio. This also prevents Asterisk from letting the upstream switch select the B-channel on outgoing calls to the pstn. Asterisk is written this way since it uses the audio channel as the fundamental unit, with the D-channel as carrier of signalling for the individual B-channels. Another way to view ISDN is to consider the D-channel the fundamental unit, which can carry several audio streams as a side effect of the signalling. The first viewpoint resembles the traditional view of telephony as individual circuits, the second resembles the ISDN/SS7 view of the world. Changing Asterisk to be more ISDN-like is quite a lot of work. Peter - Original Message - From: Arik Funke [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, August 20, 2005 7:44 PM Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the first two specified phones ring. Etc. with the first channel 21 ist called, with the second 22. How can I get asterisk to signal to all phones with just one isdn channel? I am trying to duplicate the setup I had with my old isdn pbx with did above trick just fine... Maybe somebody can help me configure asterisk appropriately? Cheers, Arik PS: I gave following a try but without success: Dial(Zap/g1/21-29,,t) Dial(Zap/g1/21+29,,t) ___ 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 Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ 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] need provider with these did's avail: (anybody?)
obviously i'm looking for a direct connect w/ asterisk (SIP or IAX), so no proprietary equip, but if you provide, or know of a provider that has any of these available, please let me know. (Tennessee) 423-869- (Kentucky) 606-337- 606-248- 606-242- ___ 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] Installing to a prefix.
Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM: Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 19 August 2005 21:27, Kevin Walsh wrote: I'll send the modified Makefiles to anyone who needs them. May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. How about submitting a disclaimer to Digium for the modified makefiles? Sincerely, Trevor Hammonds ___ 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] SPA-2100 Analog Telephone Adapter
Eric Wieling aka ManxPower wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box No, you would ignore the LAN port. When I am at home I use this setup: Phones - 2100 FXS ports - 2100 WAN port - Ethernet Switch - Asterisk If I were to get another 2100 would I use the LAN port to connect to it? Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!
On Sat, 20 Aug 2005, Paul Hewlett wrote: This does nothing but tells you what would happen. If your modprobe.d/zaptel file is correct the the output from this command will be loading of zaptel,wcfxs and an execution of ztcfg. In other words you do not have to modprobe more than one module - dependencies are sorted by the modprobe.d/zaptel file. If you want -vv on the ztcfg file edit modprobe.d/zaptel. I remember from the wiki somewhere that one must not execute ztcfg more than once and this will happen if you modprobe zaptel and then wctdm and then execute ztcfg manually. So to load Personally I always immediately remove the automatic call to ztcfg from /etc/modules.d/zaptel or modprobe.conf. Digium's boards don't mind about multiple calls to ztcfg, but the Junghanns board silently stops working and I have wasted plenty of time trying to find out why my customer's Junghanns doesn't work. So I get rid of the secret call to ztcfg and I know where I stand. Steve ___ 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] Where did my DID's go??
Your only recourse is to get your new carrier to realize that the numbers have been released and to proceed with the porting despite the fact that they have not received the notification. Thanks for the info! I've forwarded your message to the new carrier in hopes that they'll be able to do something. I've sent them a message and still no action - is there anything I can do in the interim other than deflect complaints from family, friends and system users?? Hatton ___ 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] Sudenly unable to get incoming from Broadvoice
This is very odd. I've been able to fix the problem by adding a DID route as follows exten = 9738281625,1,Dial(SIP/2208) Without this line it doesn't work. I've even rolled back from the latest CVS head to the release 1.0.8 and still it don;t work. I'm flumaxed!! Mark Mark Phillips wrote: I think I spotted what's going on. When I sift through the sip debug I see that my server is looking for my number in the context I take the calls into. Problem is its not there. I was expecting * to dump the call into the exten=s,1,blahblahblah logic but it's not doing that. Now I think about it I did a cvs code update/co,pile etc about 5 days ago. Have I discovered a newly intor'd bug? I'm gonna roll back and see if it goes away. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Help needed receiving incoming calls.
Hi All,
I've got Asterisk working and am trying to configure with Sipgate. I can make out going calls. Incoming calls show up on the AMP panel with the trunk showing red. However, the call does not go to the extension.
I initally configured Asterisk by editing the config files. I have followed the various guides and have edited sip.conf and extensions.conf copy as below.
When I then configured x-lite...nothing worked. I then went into AMP setup and used the GUI to set things up. I set up an extension (not called Xlite and trunk and DID etc. I did not delete the bits added to the config files. My trunk settings are as below.
Questions...1. Why did the editing of the config files not work?2. Why did I have to go into the GUI to set it up?3. Why does the trunk show an incoming call that is not being forwarded to the extension.4. When I set up the trunk, I got a second extension showing in the extensions part of the GUI with the Extension title '92 ( sip )' with a user name of '3141217'. The second extension shows the settings I put in the incoming trunk section. Why?
Any help would be gratefully received.
Thanks
Brian.
***
sip.conf
***[general]
port = 5060bindaddr = 0.0.0.0 disallow=allallow=gsmallow=ulawallow=alawcontext = from-sip-externalcallerid = Unknownexternip=***.***.***.***localnet=192.168.0.1localmask=255.255.255.0nat=yesregister = 3141217:[EMAIL PROTECTED]/3141217
#include sip_nat.conf#include sip_custom.conf#include sip_additional.conf
[sipgate]type=friendusername=3141217secret=passwordhost=sipgate.co.ukfromuser=3141217fromdomain=sipgate.co.uknat=yesqualify=yesauthuser=3141217dtmfmode=infocontext=incomingsipgateinsecure=verycanreinvite=nodisallow=allallow=ulawallow=alaw
[xlite1]type=friendusername=xlite1callerid= Brians notebook 201host=dynamicnat=yescanreinvite=nodisallow=allallow=ulawallow=alaw
***
extensions.conf
***
have added the following to inbound context
[incomingsipgate] exten = h,1,Hangup exten = 3141217,1,Dial(SIP/internestelefon,20,tr)
[sipgate] exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _9.,2,Playback(invalid) exten = _9.,3,Hangup
***
SIP Trunk part in GUI
Outbound caller ID: 3141217Maximum channels:
Outgoing Dial Rules:Outgoing Settings
Trunk Name: brighton outgoingPEER Details:
host=sipgate.co.uksecret=passwordtype=peerusername=3141217
Incoming Settings
User Context: 3141217User Details:
callerid=3141217context=from-pstndtmfmode=infofromdomain=sipgate.co.ukhost=sipgate.co.ukinsecure=verysecret=passwordtype=useruser=3141217username=3141217
Registration String: 3141217:[EMAIL PROTECTED]/3141217
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RE: [Asterisk-Users] Installing to a prefix.
Trevor G. Hammonds [EMAIL PROTECTED] wrote: Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM: Andrew Kohlsmith [EMAIL PROTECTED] wrote: May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. How about submitting a disclaimer to Digium for the modified makefiles? Let's not bring that subject up again. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Installing to a prefix.
On Sat, Aug 20, 2005 at 02:57:50AM +0100, Kevin Walsh wrote: Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 19 August 2005 21:27, Kevin Walsh wrote: I'll send the modified Makefiles to anyone who needs them. May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. INSTALL_PREFIX/DESTDIR is not instended for that. It is intended for installing asterisk to a different prefix than the one you build it to. It is commonly used for building installation packages (e.g: for rpms or debs). It should break if you have any hard-wired path that is set according to INSTALL_PREFIX. However PREFIX is originally intended for a package-per-directory pathers, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Where did my DID's go??
You can file a complaint with the (insert regulator body here)... You can call your provider every 5 minutes complaining bout the situation... Other than that... no.. Regards, Derek - Original Message - From: C. Hatton Humphrey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 20, 2005 3:36 PM Subject: Re: [Asterisk-Users] Where did my DID's go?? Your only recourse is to get your new carrier to realize that the numbers have been released and to proceed with the porting despite the fact that they have not received the notification. Thanks for the info! I've forwarded your message to the new carrier in hopes that they'll be able to do something. I've sent them a message and still no action - is there anything I can do in the interim other than deflect complaints from family, friends and system users?? Hatton ___ 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
Re: [Asterisk-Users] Where did my DID's go??
You can file a complaint with the (insert regulator body here)... You can call your provider every 5 minutes complaining bout the situation... Other than that... no.. Regards, Derek - Original Message - From: C. Hatton Humphrey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 20, 2005 3:36 PM Subject: Re: [Asterisk-Users] Where did my DID's go?? Your only recourse is to get your new carrier to realize that the numbers have been released and to proceed with the porting despite the fact that they have not received the notification. Thanks for the info! I've forwarded your message to the new carrier in hopes that they'll be able to do something. I've sent them a message and still no action - is there anything I can do in the interim other than deflect complaints from family, friends and system users?? Hatton ___ 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
Re: [Asterisk-Users] XML Revisited
Hi Anton, I recently contacted polycoms tech support asking if their phones supported XML pushed information to which they replied that only model 600 had a microbrwoser capable of reading dhtml files and such. My question to the community is: is somebody doing any XML info push to any brand of phones except Cisco? How are you doing it? One of the wonders of VoIP should be the means to send information back to the phone which ould be displayed on those wonderful screens that they have :) besides showing callerid and time which normal phones do.. Any ideas/comments? I wish I had a Polycom 600 to try. ;) Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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: {Scanned} RE: [Asterisk-Users] [OT] Looking for Web based SIP endpoint
On Sat, 2005-08-20 at 13:11 -0600, Tom wrote: Pulu Anau wrote: The java client you mention states on it's webpage it has to install a local .dll/.so and that it only works for x86 Windows or Linux. Does anyone know of one that's completely in Java? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Saturday, August 20, 2005 4:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] Looking for Web based SIP endpoint Kevin Walsh wrote: Matt Riddell [EMAIL PROTECTED] wrote: Jason Becker wrote: https://sip-communicator.dev.java.net/ Don't know the current state of functionality with Asterisk. I couldn't get it to work many months ago - even with help from the developer. Any reason you are looking for SIP and not IAX? Is there an IAX alternative that you'd recommend? Mozilla/FireFox:(PC/Linux) http://moziax.mozdev.org/ Java (Mac/PC/Linux): http://www.hem.za.org/jiaxclient/ ActiveX: (PC - IE) http://www.geocities.com/babarnazmi/index2.htm Think that should just about cover your bases. :) Try this site http://www.microappliances.com the only problem is that it uses IE(active x) http://cockatoo.mozdev.org/ Tom -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ 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] 1 server vs. 2 server config
We are looking at moving more of our business to distant locations. Im looking at two different network configurations and would like some thoughts or comments. Scenario 1: -We take our existing T1 route it without any conversion etc. trans-Atlantic to remote site via carrier Y -at remote site, we plug in an E1 to our asterisk server and run g7.29a or GSM internally Scenario 2: -We take our existing T1 route it with compression trans-Atlantic to remote site via carrier Y -Carrier Y un-compresses E1 -at remote site, we plug in an E1 to our asterisk server and run g7.29a or GSM internally Scenario 3: -We take our existing T1, terminate it in the States -Route voice traffic to remote site via carrier Ys data network and IAX -at remote site, we take IAX and convert to g7.29a or GSM I like #3 because it puts more control here, but Im guessing it would also add the greatest amount of latency and wed have a larger hardware cost. Thanks! -Paul ___ 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] Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
On Wed, Aug 03, 2005 at 11:28:19AM -0500, [EMAIL PROTECTED] wrote: 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1 (Gavin Hamill) Date: Wed, 3 Aug 2005 15:32:48 +0100 From: Gavin Hamill [EMAIL PROTECTED] Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii All the messages I've read on this are from people experiencing these errors in quiet times - I get them as soon as I plug a port on our TE410P to an Inter-Tel AXXESS PBX.. and I get them continuously... I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) and the PBX.. and whilst the telco ISDN30e side works like a charm [1] I simply can't get a reliable link to the PBX.. I've tried two different T1 crossovers (1-4, 2-5) with identical results and zapata.conf is indeed using signalling=pri_cpe for the telco ISDN30e and pri_net for the PBX Digium support have taken me through loopback testing which came out perfect, and the card is not sharing any IRQ, yet this error renders the card useless :( Digium are reluctant to accept a return and replace the card since they don't believe it to be at fault - and neither do I. I see the same behaviour with 1.0.9 asterisk / libpri and 1.0.9.1 zaptel... and CVS-HEAD versions of everything. Try the 1.0.7 zaptel drivers with the 1.0.9 asterisk. Works fine here. Scary but true. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] SPA-2100 Analog Telephone Adapter
Sean Rima wrote: Eric Wieling aka ManxPower wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box No, you would ignore the LAN port. When I am at home I use this setup: Phones - 2100 FXS ports - 2100 WAN port - Ethernet Switch - Asterisk If I were to get another 2100 would I use the LAN port to connect to it? You would only use the LAN port if you wanted the device to provide NAT translation/routing between the LAN port and the WAN port. ___ 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] Problems with Asterisk(*): Not-Registered
Please attach your sip.conf file also Joshua Abbott wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, Currently we have a server setup for Asterisk(*) and a TFTP server. My extension has been setup with Asterisk(*) and downloads the information from the TFTP server correctly (I've tried to erase my phone using *468 to see if that will do the trick). I have a Polycom SoundPoint 600 phone. Attached in zip format are the .cfg and .log files downloaded by the phone from the TFTP server. Still, the phone will not register with Asterisk(*). Joshua -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDB9oTeYokiwGEZc4RAhA5AJ9GgsUJfIcjSnB71gMqa7BD+DMF1ACeOzvQ Aa75CYdCnJD8Aj7VkIO+CmA= =WdOr -END PGP SIGNATURE- ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Asterisk aborts = undefined symbol: pri_channel_bridge
I have grabbed the latest zaptel, libpri, asterisk, and asterisk-addons from CVSHEAD. Everything complies and installs well, but when I go to run asterisk it aborts with: [chan_zap.so]Aug 20 21:18:53 WARNING[11840]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_channel_bridge Aug 20 21:18:53 WARNING[11840]: loader.c:543 load_modules: Loading module chan_zap.so failed! computer:# Ouch ... error while writing audio data: : Broken pipe From asterisk/messages Aug 20 21:18:53 WARNING[11840] loader.c: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_channel_bridge Aug 20 21:18:53 WARNING[11840] loader.c: Loading module chan_zap.so failed! Anyone else encounter this? Solution? ___ 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] 1.0.9 - can't get link up, 1.0.7 works fine.
Last tuesday I moved the asterisk server from 1.0.7 to 1.0.9, while leaving the zaptel drivers at 1.0.7 because it was a lunchtime update. This is a box with two TE405Ps in it, and all eight ports in use. Today I unloaded the 1.0.7 drivers and replaced them with 1.0.9 and oh boy... two of the 8 PRIs didn't want to come back, I got a million of FCS errors over the console and I got three new messages in /var/log/messages: Aug 21 11:32:16 Found a Wildcard: Wildcard TE410P/TE405P (1st Gen) [...] Aug 21 11:32:16 VPM: Not Present [...] Aug 21 11:35:11 HDLC Receiver overrun on channel TE4/1/3/16 (master=TE4/1/3/16) [repeated several million times] The PRIs didn't come back, pri show span showed them as Provisioned, Down, Active: PRI[3]: expected Status: Provisioned, Up, Active, got Status: Provisioned, Down, Active. PRI[7]: expected Status: Provisioned, Up, Active, got Status: Provisioned, Down, Active. I rebooted the box afterwards (just to get rid of gremlins), I rebooted the box on PRI 3 (PRI 7 is connected to the telco so no rebooting there) and it stayed the same. Unloading the drivers and going back to the 1.0.7 drivers with the 1.0.9 asterisk resolved everything and now it's working again. Who has suggestions for me on what to do? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Installing to a prefix.
Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Aug 20, 2005 at 02:57:50AM +0100, Kevin Walsh wrote: Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 19 August 2005 21:27, Kevin Walsh wrote: I'll send the modified Makefiles to anyone who needs them. May I humbly request they be attached to a feature request on Mantis? I've been less than humbly requested not to do that sort of thing any longer, as I haven't signed a disclaimer. Sorry about that. The Asterisk change is trivial; Just set the INSTALL_PREFIX variable in the Makefile and then modify asterisk.conf and possibly musiconhold.conf. The Zaptel Makefile changes are a bit more involved. The diff file is 148 lines long. I've never had cause to look at libpri. INSTALL_PREFIX/DESTDIR is not instended for that. It is intended for installing asterisk to a different prefix than the one you build it to. It is commonly used for building installation packages (e.g: for rpms or debs). That's fine for Asterisk. The Zaptel Makefile, as distributed, doesn't play nicely if you change the prefix. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Looking for Provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I currently have internet service through MediaCom (Cable Internet) and need to find a VOIP provider that is compatible with Asterisk and Cable Internet. Any ideas? I'm in Missouri about 1.5 hours west of St Louis, MO in a town called Hermann (65041 zip code) Joshua -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDB+6deYokiwGEZc4RAmjiAKCXEqn/X4b8blsXsqF7YM4//9COqACgp2Of IUOztI66i882+yCncoSYALk= =L8Pk -END PGP SIGNATURE- begin:vcard fn:Joshua Abbott (Successful Hosting) n:Abbott;Joshua org:Successful Hosting;Support adr:3009 Avenue J;;Attn: Joshua Abbott;Brooklyn;NY;11210;USA email;internet:[EMAIL PROTECTED] title:Technical Support Representative tel;work:+1 (866) 494-5096 x1207 tel;fax:+1 (419) 858-3241 note:Alt E-Fax: (801) 217-1123 x-mozilla-html:FALSE url:http://www.successfulhosting.com/ version:2.1 end:vcard ___ 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] IM patch
Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send 405 method not allowed to sender. I use polycom ip300. Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.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] [Asterisk-Dev] IM patch
Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send 405 method not allowed to sender. I use polycom ip300. Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1
On 7/7/05, Adam Dobrin [EMAIL PROTECTED] wrote: Also, around the same time, I isolated the IRQ that my zaptel cards were on. (so neither zaptel card shared its IRQ). you can see what IRQ's are in use with lspci -vb This is more likely to be the cause of the fix. Adam Dobrin wrote: Lance, I was in a similar situation, though i was rec'ing the event 6 message, i noticed no degradation of sound and so ignored it. I've since removed a *load* of unused modules, and it appears that the message is no longer coming in. I had read that some people were only getting the message after the machine had been up for a few days.. I'll check back then. This is what i added to modules.conf: noload = res_musiconhold.so noload = pbx_wilcalu.so noload = app_image.so noload = app_url.so noload = app_adsiprog.so noload = app_getcpeid.so noload = app_milliwatt.so noload = app_zapateller.so noload = app_festival.so noload = app_lookupblacklist.so noload = app_random.so noload = app_ices.so noload = app_nbscat.so noload = app_zapras.so noload = codec_adpcm.so noload = cdr_sqlite.so Adam, Well, I set up all these modules to not be loaded, and a few others, and it still happenes. I will try playing with the IRQs but I only have one digium card in my box. I was wondering, did you set your cards to not share IRQs with any other device or just not share with each other? Thanks, -- Thanks, Lance Grover ___ 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
