RE: [Asterisk-Users] X100P fails to detect user hung up
[EMAIL PROTECTED] wrote: Ron, It is a multi-reported problem, yet no resolution. I would suggest it is a bug. I have had intermittent success with POTS provided by AllTel in Texas. My opinion, you're SOL and there is very little you can do. I keep hoping that someone at digium will pick up on this and look at the hardware design etc. BTW, I tried kewlstart, loopstart etc. and it doesn't make any difference. As I said, it's intermittent on POTS, and it's constant on my ISDN fxs channels. Cheers, Willy Have you reported this issue as a bug? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk QSIG
Hello, Does anybody know if Asterisk can support QSIG protocols to be interconnected with a Traditionnal PABX? (Using a HFC chipset based ISDN card to emulate NT Interface) Thank you in advance ;-) Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Q.SIG
Hello, Does anybody know if Asterisk can support QSIG protocols to be interconnected with a Traditionnal PABX? (Using a HFC chipset based ISDN card to emulate NT Interface) Thank you in advance Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice versus data T1s: Balance of power
I hope this question isn't flamebait. I don't know anything about voice T1. What are the tradeoffs in terms of asterisk's design and performance whether traffic is handled by one type or the other? I wonder about the economics, too. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
What version of the Phone firmware are you running ? I had the same problem until I upgrade to 1.0.4.54 Chris - Original Message - From: pesb [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 24, 2004 9:41 PM Subject: Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP Hi there, I am still trying to make the asterisk SIP proxy server work with my Grandstream 100 IP phones. I tried Stephen advice and it did not work. I stil got the 404 error message. So, rigth now, I am trying the following configuration(sip.conf): ### ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT ;localnet = 192.168.0.0 ; Internal NETWORK address ;localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw ;allow=ilbc ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 1234 here. ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ;mailbox=1234,2345 ; Mailbox for message waiting indicator ;restrictcid=yes ; To have the callerid restriced - sent as ANI ;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic ;qualify=1000 ; Consider it down if it's 1 second to reply ;callgroup=1,3-4 ;pickupgroup=1,3-4 ;defaultip=192.168.0.60 ;[cisco] ;type=friend ;username=cisco ;secret=blah ;nat=yes ; This phone may be natted ;host=dynamic ;canreinvite=no ; Cisco poops on reinvite sometimes ;qualify=200 ; Qualify peer is no more than 200ms away ;defaultip=192.168.0.4 ;[cisco1] ;type=friend ;username=cisco1 ;fromuser=markster ; Specify user to put in from instead of callerid ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode tp ease billing [1001] type = friend context = default secret = gol host = dynamic callerid = STREAM-1001 1001 ;dtfmmode=inband canreinvite=no defaultip=192.168.0.105 [1002] type = friend context = default secret = gol host = dynamic callerid = STREAM-1002 1002 ;dtfmmode=inband canreinvite=no defaultip=192.168.0.104 ## This is the configuration of my SIP-phones: ipaddr=192.168.0.105 sipserver=192.168.0.102 sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=1001 authenticateid=1001 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no send_dtmf=in-audio dtmf_payload_type=101 time_zone=GMT-0 ipaddr=192.168.0.104 sipserver=192.168.0.102 sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=1004 authenticateid=1004 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no send_dtmf=in-audio dtmf_payload_type=101 time_zone=GMT-0 What's wrong here?? When I try to dial from one phone to the other, I get 404 error message. Please, somebody help me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Program to manage the faxes
Hello, A few weeks I wrote a message about a program to manage faxes. A few people responds which I appreciate. For this program I made a project site on http://www.sourceforge.net. The project page for the fax program is: http://tafm.sourceforge.net If you would like to download the program directly go here: https://sourceforge.net/project/showfiles.php?group_id=105174package_id=113216 Greetings and good luck, Johan Hollemans Hello, For our company we will use Asterisk to receive our faxes. We would like to manage this faxes. To give some information to a fax. Here fore I wrote a little application. To manage the incoming faxes I wrote a script which checks the directory where the faxes are stored. If there is a new fax, this fax is send to your email address. Also there is a web interface which gives you the possibility to manage the fax. For each fax you can give some information to it. A list is given with the incoming faxes also a list with the last ten faxes is shown. And you can browse in an archive. Documentation is included in the faxprogram.tar.gz. The .doc and .sxw documentation are the same. Read this documentation which will gives you information how the program works and what you have to change to get it work. I hope you can use this program for managing your faxes. If there are any bugs, questions or you have some modifications or new features let me know. Excuse me if my English is not good. Greetings, Johan Hollemans from Synantics B.V. The Netherlands ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 International Termination
Thanks for your feedback. We will look into it. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Thursday, March 25, 2004 2:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 International Termination On Thu, 25 Mar 2004, Anton Tinchev wrote: Some troubles with dtmf sending. I tested here (I'm preparing a report to send to support at diamondcard dot us) and I found that they only support dtmfmode=info. Before I was using dtmfmode=rfc2833. Using a Cisco 7960G phone. I don't know if this explains your problem... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
Well ... For starters, in your sip.conf you have dtmfmode=rfc2833 but your phone setup gives send_dtmf=in-audio In your post (below) you also left out authenticate_password=gol but that may be an oversight? BTW: My GS setup uses dtmfmode=info (in my sip.conf for each phone) and send_dtmf=SIP_IPNFO in the phone config Cheers, Willy - Original Message Follows - Hi there, I am still trying to make the asterisk SIP proxy server work with my Grandstream 100 IP phones. I tried Stephen advice and it did not work. I stil got the 404 error message. So, rigth now, I am trying the following configuration(sip.conf): ### ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT ;localnet = 192.168.0.0 ; Internal NETWORK address ;localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw ;allow=ilbc ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 1234 here. ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ;mailbox=1234,2345 ; Mailbox for message waiting indicator ;restrictcid=yes ; To have the callerid restriced - sent as ANI ;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic ;qualify=1000 ; Consider it down if it's 1 second to reply ;callgroup=1,3-4 ;pickupgroup=1,3-4 ;defaultip=192.168.0.60 ;[cisco] ;type=friend ;username=cisco ;secret=blah ;nat=yes ; This phone may be natted ;host=dynamic ;canreinvite=no ; Cisco poops on reinvite sometimes ;qualify=200 ; Qualify peer is no more than 200ms away ;defaultip=192.168.0.4 ;[cisco1] ;type=friend ;username=cisco1 ;fromuser=markster ; Specify user to put in from instead of callerid ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode tp ease billing [1001] type = friend context = default secret = gol host = dynamic callerid = STREAM-1001 1001 ;dtfmmode=inband canreinvite=no defaultip=192.168.0.105 [1002] type = friend context = default secret = gol host = dynamic callerid = STREAM-1002 1002 ;dtfmmode=inband canreinvite=no defaultip=192.168.0.104 ## This is the configuration of my SIP-phones: ipaddr=192.168.0.105 sipserver=192.168.0.102 sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=1001 authenticateid=1001 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no send_dtmf=in-audio dtmf_payload_type=101 time_zone=GMT-0 ipaddr=192.168.0.104 sipserver=192.168.0.102 sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=1004 authenticateid=1004 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no send_dtmf=in-audio dtmf_payload_type=101 time_zone=GMT-0 What's wrong here?? When I try to dial from one phone to the other, I get 404 error message. Please, somebody help me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 International Termination
This should be taken off the list -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Wednesday, March 24, 2004 8:37 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 International Termination On Thu, 25 Mar 2004, Anton Tinchev wrote: Some troubles with dtmf sending. I tested here (I'm preparing a report to send to support at diamondcard dot us) and I found that they only support dtmfmode=info. Before I was using dtmfmode=rfc2833. Using a Cisco 7960G phone. I don't know if this explains your problem... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Can you try: http://yourdomain/asterisk-stat/graph_stat.php?min_call=fromstatsday_sday=25days_compare=2fromstatsmonth_sday=2004-03 And tell me about the result ! -Areski On Thu, 2004-03-25 at 11:13, Robert Boardman wrote: Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Hi Areski it comes back with a blank page? Robb Areski [EMAIL PROTECTED] said: Can you try: http://yourdomain/asterisk-stat/graph_stat.php?min_call=fromstatsday_sday=25days_compare=2fromstatsmonth_sday=2004-03 And tell me about the result ! -Areski On Thu, 2004-03-25 at 11:13, Robert Boardman wrote: Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sched_settime error
I have the following problem in playback: When any sound file is played back, it is garbled for a few seconds and the following error displays: Sched_settime: Request to schedule in the past? After about 5 seconds, the sound clears up and the error stops. What gives???
RE: [Asterisk-Users] CDR and Mysql (or Postgre)
Download asterisk-addons from the CVS. Compile it the same way you compile asterisk and it's other modules. Make sure you have MySQL installed and running. Then read the file /usr/src/asterisk-addons/doc/cdr_mysql.txt for information on how to create the necessary tables in your database. The last step is to copy the config file: /usr/src/asterisk-addons/cdr_mysql.conf.sample to /etc/asterisk/cdr_mysql.conf and edit it with the username, database, password, etc. for your particular environment. Its really that simple, and all calls are then inserted into the database when they end. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Wednesday, March 24, 2004 10:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CDR and Mysql (or Postgre) I have been researching using mysql as a database to manage the cdr's. However, I do not see how to get asterisk to insert the records directly in the database. All I can see from my searches is some scripts to copy the master.csv text file into the database. However I think this would be problematic. If you copy at midnight, then do you erase the file after you copy it? What about calls that are still ongoing? Where do they get logged? How are other people moving the CDR's into a database in real time for billing either pre-paid or post-paid? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.614 / Virus Database: 393 - Release Date: 3/5/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.614 / Virus Database: 393 - Release Date: 3/5/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Detection On incoming calls
I've got a single inbound analogue line setup with 2 phone numbers and distinctive ring and I'm trying to setup distinctive ring detection to separate calls and put a distinctive ring to the extensions based on what number was called... Problem is it seems most countries send a distinctive ring then the caller ID, however here it appears a short ~50ms ring is sent, followed by a pause with caller ID *then* the proper ring/distinctive ring is sent, is there any simple way to get asterisk to ignore trying to match a distinctive ring with the first 50ms segment, and do it on the 2nd segment instead? Needless to say it was showing up as 0,0,0 ever time no matter which phone number was called... Both myself and a friend have tried coding in methods to shorten rings and flags to try and make asterisk try to set a context for the call on the first ring but this doesn't seem to work and it still gets passed off... Any help would be greatly appreciated... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Sounds like it s missing smth in your php conf ! have a look to this good tutorial http://www.zend.com/zend/tut/tutsweat3.php Check your configuration with the conf information provided there and then try to make working a jpgraph sample on your server... Hope that it will help, Regards,Areski On Thu, 2004-03-25 at 13:18, Robert Boardman wrote: Hi Areski it comes back with a blank page? Robb Areski [EMAIL PROTECTED] said: Can you try: http://yourdomain/asterisk-stat/graph_stat.php?min_call=fromstatsday_sday=25days_compare=2fromstatsmonth_sday=2004-03 And tell me about the result ! -Areski On Thu, 2004-03-25 at 11:13, Robert Boardman wrote: Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some bugs, cause it was not possible to choose criteria and then browse the result page by page... now it's work fine :) So, better to make an update of your version http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_2.tar.gz http://www.areski.net/asterisk-stat-v1/about.php Sorry about all this changes... Regards, Areski On Wed, 2004-03-24 at 16:42, Ryan Thrash wrote: Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
Dear Chris, My firmware version is 1.0.4.39, how can I make the upgrade? where (url site) can I get the firmware? thanks again, Pablo S. On Thursday 25 March 2004 06:32, Chris Stenton wrote: What version of the Phone firmware are you running ? I had the same problem until I upgrade to 1.0.4.54 Chris - Original Message - From: pesb [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 24, 2004 9:41 PM Subject: Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP Hi there, I am still trying to make the asterisk SIP proxy server work with my Grandstream 100 IP phones. I tried Stephen advice and it did not work. I stil got the 404 error message. So, rigth now, I am trying the following configuration(sip.conf): ### ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT ;localnet = 192.168.0.0 ; Internal NETWORK address ;localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw ;allow=ilbc ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 1234 here. ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ;mailbox=1234,2345 ; Mailbox for message waiting indicator ;restrictcid=yes ; To have the callerid restriced - sent as ANI ;[pingtel] ;type=friend ;username=pingtel ;secret=blah ;host=dynamic ;qualify=1000 ; Consider it down if it's 1 second to reply ;callgroup=1,3-4 ;pickupgroup=1,3-4 ;defaultip=192.168.0.60 ;[cisco] ;type=friend ;username=cisco ;secret=blah ;nat=yes ; This phone may be natted ;host=dynamic ;canreinvite=no ; Cisco poops on reinvite sometimes ;qualify=200 ; Qualify peer is no more than 200ms away ;defaultip=192.168.0.4 ;[cisco1] ;type=friend ;username=cisco1 ;fromuser=markster ; Specify user to put in from instead of callerid ;secret=blah ;host=dynamic ;defaultip=192.168.0.4 ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode tp ease billing [1001] type = friend context = default secret = gol host = dynamic callerid = STREAM-1001 1001 ;dtfmmode=inband canreinvite=no defaultip=192.168.0.105 [1002] type = friend context = default secret = gol host = dynamic callerid = STREAM-1002 1002 ;dtfmmode=inband canreinvite=no defaultip=192.168.0.104 ## This is the configuration of my SIP-phones: ipaddr=192.168.0.105 sipserver=192.168.0.102 sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=1001 authenticateid=1001 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no send_dtmf=in-audio dtmf_payload_type=101 time_zone=GMT-0 ipaddr=192.168.0.104 sipserver=192.168.0.102 sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=1004 authenticateid=1004 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no send_dtmf=in-audio dtmf_payload_type=101 time_zone=GMT-0 What's wrong here?? When I try to dial from one phone to the other, I get 404 error message. Please, somebody help me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To
Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
On Thu, 2004-03-25 at 12:50, pesb wrote: My firmware version is 1.0.4.39, how can I make the upgrade? where (url site) can I get the firmware? http://www.grandstream.com/BETATEST/ -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register Asterisk
Necessary to create a register in the Asterisk, more it has that to send the information: username, password, sip proxy, outboundproxy, domain/real. Help to decide this problem me? ThankĀ“s Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device
I have tried to connect asterisk (which I use through hisax isdn4linux device) with mediatrix sip device with g729 codec asterisk can not connect with mediatrix (it connects when ulaw/alaw are used) when g729 is forced any ides what to do? Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 178 Content-Type: application/sdp Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, REFER v=0 o=MxSIP 0 0 IN IP4 192.168.3.211 s=SIP Call c=IN IP4 192.168.3.211 t=0 0 m=audio 5004 RTP/AVP 18 8 0 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 10 headers, 9 lines Found audio format UNKN Found audio format ALAW Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Capabilities: us - 268, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: sip:[EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 Mar 25 15:24:26 NOTICE[1225991360]: channel.c:1513 ast_set_write_format: Unable to find a path from UNKN to SLINR Mar 25 15:24:26 WARNING[1225991360]: channel.c:1920 ast_channel_make_compatible: No path to translate from Modem[i4l]/ttyI0(64) to SIP/301-3309(256) Mar 25 15:24:26 WARNING[1225991360]: app_dial.c:702 dial_exec: Had to drop call because I couldn't make Modem[i4l]/ttyI0 compatible with SIP/301-3309 set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 asterisk*CLI Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 0 7 headers, 0 lines Sometimes you're the bug, sometimes you're the windshield. mailto:[EMAIL PROTECTED] http://printel.hr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A tidbit about one-way audio ethernet aliases
Hey all, Thought I'd share a curiosity I found when trying to use heartbeat software for asterisk failover (this may already be common knowledge to some/many, but I hadn't seen mention of it yet). The default ha-linux ip-takeover script uses ifconfig to create an ethernet alias to which a secondary IP address is assigned (i.e. eth0 is your main interface at 10.1.1.1, and the heartbeat script creates eth0:0 at 10.1.1.2). I had been testing my asterisk configuration w/out heartbeat 'til I thought it stable enough for production, then I turned on the heartbeat left the office to set up my first subscriber. Imagine my shame... No audio from pstn to subscriber (using sip ata behind nat). Seems the rtp stream doesn't appreciate being directed at a secondary address. So, swapping out the default ha-linux ip-takeover script for one that uses ip from the iproute2 package solved my problem. (Perhaps this is what Doichin Dokov had going on late last week?) Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UNSUBSCRIBE
UNSUBSCRIBE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Detection On incoming calls
Duane, On Thu, 25 Mar 2004, Duane wrote: Problem is it seems most countries send a distinctive ring then the caller ID, however here it appears a short ~50ms ring is sent, followed by a pause with caller ID *then* the proper ring/distinctive ring is sent, is there any simple way to get asterisk to ignore trying to match a distinctive ring with the first 50ms segment, and do it on the 2nd segment instead? http://bugs.digium.com/bug_view_page.php?bug_id=0001007 I lodged this patch some time ago, but no action -- I think it's brutally ugly, but it worked for me. By all means, try it out (or improve on it). It would be much better to change the way the code works by controlling where the CID and/or distinctive ring detection is done via configuration, rather than just repeating a block of code like I did. I never got around to making a version two (yet). There may also be a conflict with the #define DEFAULT_CIDRINGS 2 that we require here to generate proper CID data for analogue handsets attached to zaptel FXS channels. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 International Termination
Seems to me this thread should be taken off-list. It's effectively a beta test list for a Dreamtime product, so Dreamtime should set up their own list and at most send a single invite to it to the asterisk list. Carey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P fails to detect user hung up
Try calling application Hangup at the ends of the extension chains. Works for me. Bob [EMAIL PROTECTED] wrote: Ron, It is a multi-reported problem, yet no resolution. I would suggest it is a bug. I have had intermittent success with POTS provided by AllTel in Texas. My opinion, you're SOL and there is very little you can do. I keep hoping that someone at digium will pick up on this and look at the hardware design etc. BTW, I tried kewlstart, loopstart etc. and it doesn't make any difference. As I said, it's intermittent on POTS, and it's constant on my ISDN fxs channels. Cheers, Willy - Original Message Follows - I am using the wildcard X100P with *. PSTN line comes in to the FXO port of this card. Everything works fine most of the time. However, occasionally Asterisk doesn't seem to be able to detect the user has hung up and therefore tie up the line for quite a long time. Does anyone know if there's anything I can do to fix this problem? thanks Ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device
You need a G729 license for asterisk to make a connection. You have to get them from diguim, they are $10 a channel. They do give you a single channel demo license, you just have to get it from them. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marko Rakar Sent: Thursday, March 25, 2004 8:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device I have tried to connect asterisk (which I use through hisax isdn4linux device) with mediatrix sip device with g729 codec asterisk can not connect with mediatrix (it connects when ulaw/alaw are used) when g729 is forced any ides what to do? Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 178 Content-Type: application/sdp Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, REFER v=0 o=MxSIP 0 0 IN IP4 192.168.3.211 s=SIP Call c=IN IP4 192.168.3.211 t=0 0 m=audio 5004 RTP/AVP 18 8 0 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 10 headers, 9 lines Found audio format UNKN Found audio format ALAW Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Capabilities: us - 268, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: sip:[EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 Mar 25 15:24:26 NOTICE[1225991360]: channel.c:1513 ast_set_write_format: Unable to find a path from UNKN to SLINR Mar 25 15:24:26 WARNING[1225991360]: channel.c:1920 ast_channel_make_compatible: No path to translate from Modem[i4l]/ttyI0(64) to SIP/301-3309(256) Mar 25 15:24:26 WARNING[1225991360]: app_dial.c:702 dial_exec: Had to drop call because I couldn't make Modem[i4l]/ttyI0 compatible with SIP/301-3309 set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 asterisk*CLI Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 0 7 headers, 0 lines Sometimes you're the bug, sometimes you're the windshield. mailto:[EMAIL PROTECTED] http://printel.hr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 International Termination
Hello, Thats a good idea. We will go and set up our own list for this discussion. Thanks for the suggestion. Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carey Jung Sent: Thursday, March 25, 2004 3:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX2 International Termination Seems to me this thread should be taken off-list. It's effectively a beta test list for a Dreamtime product, so Dreamtime should set up their own list and at most send a single invite to it to the asterisk list. Carey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
Sorry about the post to the wrong level of the thread, but something was wrong with the first copy of the message (i.e., my mail reader wouldn't display it). Comments are inline. I tried Stephen advice and it did not work. I stil got the 404 error [general] dtmfmode=rfc2833 This does not match the selection used in your phone, and ironically, is the only choice that does not seem to work on the GS phones. Use inband or info and make sure that you set the phone the same way. ;[snomsip] ;type=friend ;secret=blah ;[pingtel] ;[cisco] ;[cisco1] You might consider deleting all of these unused bits from your file, or at least from the email before you send it. If you need them later, you can always copy and paste them back from a reference copy of the file. [1001] type = friend context = default secret = gol host = dynamic Unless you have a good reason for using the dynamic option, I would not use it. In your case, the phone's IP is Hardwired, and private to boot. Just put the IP in after the host=. You also avoid the (possibly still present) grandstream bug which loses registrations from time to time. callerid = STREAM-1001 1001 ;dtfmmode=inband Ironically, this is what you used on the phons. Why is it commented here? canreinvite=no defaultip=192.168.0.105 [1002] Same for phone 2 This is the configuration of my SIP-phones: outboundproxy=null outboundproxy_port=null If all else fails, put your server IP in here! Use default port registration_expiration=10 You may find registration to be a problem with the GS. See comments above. send_dtmf=in-audio This must match the entry in sip.conf (In the GS world, in-audio = inband) Sincerely, Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX and Snom200
We thought about this option. I guess the IAX2 is not the problem. We believe the real problem will be the user interface. snom would have no problem providing the platform (hardware plus operating system and stuff like audio), but we simply dont want to open another development branch (already got enough trouble with SIP.-). I personally think its ok to optimize the SIP interoperability. All that you can do in IAX can also be done in SIP (or am I making a big mistake here?). Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Thursday, March 25, 2004 4:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX and Snom200 Greetings What would it take to have a snom200 support IAX, what are the processes or having hardware to support a new codec? Can this be tested and done by a uesr or must this be done by the manufacturer? Thanks in advance
[Asterisk-Users] IAX Termination
I have a problem w/ a IAX termination provider. Recently when people would call over IAX, they could hear everyting even dial an extension. When the extension picked up, I could hear them but they could not hear me. My ZAP device works fine, it's just coming in over the IAX. I updated to the latest CVS yesterday with the same result. No audio is being passed on one leg of the call. Any help would be appreciated. I have sent an email to support without much help except for upgrade to the latest cvs Here's the console: -- Accepting AUTHENTICATED call from 66.225.202.72, requested format = 4, actual format = 4 -- Executing Goto([EMAIL PROTECTED]/7, s|1) in new stack -- Goto (autoattend,s,1) -- Executing Answer([EMAIL PROTECTED]/7, ) in new stack -- Executing Wait([EMAIL PROTECTED]/7, 1) in new stack -- Executing DigitTimeout([EMAIL PROTECTED]/7, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout([EMAIL PROTECTED]/7, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround([EMAIL PROTECTED]/7, thanks) in new stack -- Playing 'thanks' (language 'en') == CDR updated on [EMAIL PROTECTED]/7 -- Executing Macro([EMAIL PROTECTED]/7, oneline|5000) in new stack -- Executing Dial([EMAIL PROTECTED]/7, SIP/5000|30|t) in new stack -- Called 5000 -- SIP/5000-18b2 is ringing -- Nobody picked up in 3 ms -- Executing VoiceMail2([EMAIL PROTECTED]/7, u5000) in new stack -- Playing 'voicemail/default/5000/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/5000/INBOX/msg0002 format: wav49, 0x8135140 -- x=1, open writing: /var/spool/asterisk/voicemail/default/5000/INBOX/msg0002 format: gsm, 0x8135368 -- x=2, open writing: /var/spool/asterisk/voicemail/default/5000/INBOX/msg0002 format: wav, 0x8135478 Mar 24 16:30:38 WARNING[1272032560]: app_voicemail.c:1260 play_and_record: No audio available on [EMAIL PROTECTED]/7?? -- User hung up == Spawn extension (macro-oneline, s, 2) exited non-zero on '[EMAIL PROTECTED]/7' in macro 'oneline' == Spawn extension (autoattend, 5000, 1) exited non-zero on '[EMAIL PROTECTED]/7' -- Hungup '[EMAIL PROTECTED]/7' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Message Extension support
I've successfully installed Asterisk and have Microsoft's Instant Messenger connecting. We can make VoIP calls between clients without a problem, however we cannot send text instant messages between clients. From what I can tell this should be possible using IETF SIMPLE or RFC 3428 (SIP Message Extension). I can't find any reference to this and asterisk. Is it supported? Here is what I get on the Asterisk console when I send a text message. Asterisk appears to receive and transmit the message to the destination but it never actually appears. Sip read: MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:14754 From: inetsup sip:[EMAIL PROTECTED];tag=97f9a632-d8a4-4eff-9870-92bd8037b421 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE Contact: sip:192.168.0.103:14754 User-Agent: Windows RTC/1.0 Content-Type: text/plain; charset=UTF-8;msgr=WAAtAE0ATQBTAC0ASQBNAC0ARgBvAHIAbQBhAHQAOgAgAEYATgA9AE0AUwAlADIAMABTAGgAZQBsAGwAJQAyADAARABsAGcAOwAgAEUARgA9ADsAIABDAE8APQAwADsAIABDAFMAPQAwADsAIABQAEYAPQAwAA0ACgANAAoA Content-Length: 4 test 10 headers, 1 lines Receiving message! Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.103:14754 From: inetsup sip:[EMAIL PROTECTED];tag=97f9a632-d8a4-4eff-9870-92bd8037b421 To: sip:[EMAIL PROTECTED];tag=as200391f2 Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.103:2990 linux*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 International Termination
Humble apologies for using list space for this. The message is actually for Stephen Karrington. I wrote a lengthy reply to you directly (Stephen), but it was bounced by your spam filter. If you are interested in seeing it, please contact me directly, and let me know how else to forward that email to you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SoftFAX/spandsp
Hi all, My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This version has a number of changes in the way the V.29 modem works. It also has some missing functionality in the T.30 implementation filled in - it was not handling EOM messages. The previous version failed for several reasons with a Dialogic VFX/40ESC. This version succeeds, although it still seems to get a few bit errors, giving some flaws on the received image. I do not see these errors with the other FAX machines I have tried. It seems like a fairly big improvement though, and work will continue to make it better. app_rxfax.c and app_txfax.c have gained a new feature. Previously they always started in answering party mode. Now this is the default behaviour, but something like: exten = 5678,1,txfax(/tmp/testfax.tif|caller) will make them start in calling party mode. So far, these two apps have been little more that testbeds for spandsp. It seems some people are trying to use them for real work, so it seems like they should be gaining more features. The caller mode option was asked for. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 0.7.2 with cisco router 7960
I've successfully installed Asterisk 0.7.2, before we used 0.4.0 than it was working well but we needed context with date/time. In 0.7.2, we have trouble when use DTMF. We make outgoing calls from CISCO 7960 and use Cisco 2621 like gateway. About audio/voice all is working right, but DTMF donĀ“t work. Allways, for example, 2010 is seen as 210 or 10. Cisco 7960 is setting avt on DTMF. In sip.conf, [general] context is dtmfmode=rfc2833 but in [cisco2600] context is inband. In extensions.conf, all outgoing call is using SIPDtmfMode(rfc2833) because if dtmf tones in inband are not recognized on other side (like IVR or PBX in PSTN) In cisco router, we have one E1 with following config : dial-peer voice 700 pots application session destination-pattern 13T port 1/0 Can anybody help us, Please? Thanks in advance Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and Snom200
- Original Message - From: Christian Stredicke To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 10:05 AM Subject: RE: [Asterisk-Users] IAX and Snom200 We thought about this option. I guess the IAX2 is not the problem. We believe the real problem will be the user interface. snom would have no problem providing the platform (hardware plus operating system and stuff like audio), but we simply donĀt want to open another development branch (already got enough trouble with SIP.-). I personally think its ok to optimize the SIP interoperability. All that you can do in IAX can also be done in SIP (or am I making a big mistake here?). Christian There is the big difference. in that IAX handles NAT much better, esp. double NAT (security) I'm not sure if you work for snom, but I'm willing to help out where I can. Anyone else care to list the differences between SIP and IAX2? If would be great to get a comprehensive list, Mark or the digium guys ??? Barry
RE: [Asterisk-Users] UNSUBSCRIBE
UNSUBSCRIBE No. I don't want to. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 and SCSI
Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: G.729 and SCSI
Sergio, Did you try to install G729 while you had a CD in the CDROM drive? Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com --__--__-- Message: 4 From: Sergio Serrano [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 25 Mar 2004 17:48:21 +0100 Subject: [Asterisk-Users] G.729 and SCSI Reply-To: [EMAIL PROTECTED] Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 and SCSI
Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing It's not specifically listed there, but the licensing process has issues with SCSI only systems. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax questions ?
Juan == Juan J Sierralta P [EMAIL PROTECTED] writes: Juan I been playing with RxFax ... I received a FAX and it seems Juan that the aspect ratio of the image is different, ... The image Juan resolution is 1728x1092. Traditional fax has two resolutions: 98 lines/inch and 196 lines/inch. Usually the former is used unless you specify the latter when sending. So, unless you specify fine mode, what you got is normal. To view it as it would print, you must either stretch it 2x vertically or view it in software like viewfax that does that for you. -JimC by Frank D. Cringle [EMAIL PROTECTED] cf ftp://ftp.sgi.com/sgi/fax/contrib/viewfax/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 March 2004 10:51 pm, Adam Hart wrote: Comment below... Steve wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 March 2004 08:45 pm, James H. Thompson wrote: No guarantee then when public IPs match that clients are both on same NAT LAN. Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP 123.123.123.123 --- Internet Client B 192.168.0.1 - NAT Router B -| The thing is that it's all controlled by your gateway configuration. This is where you define where you find what. You must know the IP (or domain name and use DNS) of where the recipient is. If you are calling a local host you must know the IP. If you call an external host you must also must know his internet address. He'd have a redirect in his firewall that would route to his internal machine. You have no need/use of knowing what his internal IP address is. I've done all the above in many combinations. I have one setup on CA and one in FL. I have had CA call over IP to FL, then fwd the call to a local external land line and call right back in again on another land line. I have called and transferred calls to a local LAN phone as well as over the Internet. I can't really follow what you're saying, the above setup is a problem with the current IAX. Put simply, when two people are behind the same NAT device and the asterisk box is outside this nat, some NAT routers can't bridge the calls so the call is forced to continue to route through the asterisk box. This is most common cause of compliant of latency for the firefly network. Sure SOME routers understand but most don't. Ah, then it's my mistake. loosly following the thread I got a different picture. NAT/Redirect is not always done correctly by the manufacturer, try a different brand. (I always build and use OpenBSD for firewalls. It needs 500MB and 48M RAM and 2 NICs. Takes 15 minutes to build and about the same to configure. Though it's not physically a tiny custom device...) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAYxLXljK16xgETzkRAv0BAKCfH9i9K4Z3sk1RQI2feKhmkkojHwCdGd4S djDCh4dZIcG2sdD0ePPu3JY= =0mDZ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 and SCSI
Yes I have mounted CDROM first with automount(/dev/cdrom) and second manually(/dev/hde) but nothing. Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Andrew Thompson Enviado el: jueves, 25 de marzo de 2004 17:59 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] G.729 and SCSI Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing It's not specifically listed there, but the licensing process has issues with SCSI only systems. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Message Extension support
Hal A. Lightwood wrote: I've successfully installed Asterisk and have Microsoft's Instant Messenger connecting. We can make VoIP calls between clients without a problem, however we cannot send text instant messages between clients. From what I can tell this should be possible using IETF SIMPLE or RFC 3428 (SIP Message Extension). I can't find any reference to this and asterisk. Is it supported? No. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 and SCSI
If memory servers, and everyone feel free to flame away if it serves badly, the library only searches hda,hdb,hdc, and hdd. Try switching where your controller is, that may solve it. Derek -Original Message- From: Sergio Serrano [mailto:[EMAIL PROTECTED] Sent: Thursday, March 25, 2004 12:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G.729 and SCSI Yes I have mounted CDROM first with automount(/dev/cdrom) and second manually(/dev/hde) but nothing. Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Andrew Thompson Enviado el: jueves, 25 de marzo de 2004 17:59 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] G.729 and SCSI Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing It's not specifically listed there, but the licensing process has issues with SCSI only systems. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP-IP
Hi, Im new in Asterisk world. Could Somebody tell me if Asterisk solution is cpable to give IP-IP PBX service ? If the answer is yes, wich is the moodule o service name? Thanks Mariano __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX and Snom200
Certainly there is the NAT issue and this should not be underestimated. Also IAX allows optimisation of existing bandwidth between Asterisk servers. The SNOM guys should look over their shoulders at Verbiage who are bringing an IAX phone to market. I suspect it will have a lot of interest amongst this community. Brian -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Barry FawthropSent: 25 March 2004 16:07To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] IAX and Snom200 - Original Message - From: Christian Stredicke To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 10:05 AM Subject: RE: [Asterisk-Users] IAX and Snom200 We thought about this option. I guess the IAX2 is not the problem. We believe the real problem will be the user interface. snom would have no problem providing the platform (hardware plus operating system and stuff like audio), but we simply donĀt want to open another development branch (already got enough trouble with SIP.-). I personally think its ok to optimize the SIP interoperability. All that you can do in IAX can also be done in SIP (or am I making a big mistake here?). Christian There is the big difference. in that IAX handles NAT much better, esp. double NAT (security) I'm not sure if you work for snom, but I'm willing to help out where I can. Anyone else care to list the differences between SIP and IAX2? If would be great to get a comprehensive list, Mark or the digium guys ??? Barry
RE: [Asterisk-Users] IAX2 as an IETF Standard?
Here's a recap of what I am hearing: 1) Everybody (thus far) is in favor of trying to standards-track (or at least do an Information RFC) on IAX2. 2) IAX2 needs to have AES encryption added prior to submission. 3) IAX2 needs to have non pin-wheeling NAT support added (i.e. support for intra-NAT operations with an external or extra-NAT located server). 4) We need to find somebody or a committee of people who can take the time to write up the RFC and (perhaps) research the process of submitting the RFC for standards tracking. Here's my additional request for the protocol prior to finalization/formalization: 5) IAX2 needs to be able to optionally limit an IAX user/login to a single session. I have written this up in great detail at: http://bugs.digium.com/bug_view_page.php?bug_id=0001164 I think this is necessary from a network/user management standpoint. Has anybody spoken with Mark about this? I know he was thinking in terms of creating an RFC some time ago, and as leader and owner of the copyright on the prime implementation of IAX2, we could use his input and blessing. Here is a link to the IETF's RFC on submitting RFCs: http://www.ietf.org/rfc/rfc2026.txt Here is a link to the IPTel working group home page: http://www.softarmor.com/iptel/ It looks like we have to create an Internet Draft which is assigned to the relevant working group for revision, questions, comments, more revision, then it may or may not become an RFC. Unfortunately, the IPTel working group appears to be made up of people who are heavily invested in SIP. Is anybody currently active on any of the working groups in the IETF? Does anybody know anybody in the IPTel group? That may help us get started. Thoughts? grumbleOff Topic - WHY does the IETF insist on using pre-paginated text files as the official RFC format? What a HORRIBLE format for documents.../grumble Thanks, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P to E100P for stress test
Dear i have two box and i want made some stress test with one TE410P and a E100P with only one span 1 Server TE410P Span1-- PBX Span2---E100P Box The Box with TE410P is Mandrake 9.2 with P4 HT #zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow span=3,0,0,ccs,hdb3,crc4,yellow span=4,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_cpe context=prepaid immediate=no callerid=asreceived channel = 1-15,17-31,63-77,79-93 ; ,125-139,141-155,187-201,203-217 group = 2 switchtype = euroisdn ;signalling = pri_cpe ;signalling = pri_net context=demo ;immediate=yes channel = 32-46,48-62,94-108,110-124 ; ,156-170,172-186,218-232,234-248 The second Box E100P Mandrake 9.2 P3 800Mhz #zaptel.conf /// Add PRI T100P span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_net context=incoming immediate=yes callerid=asreceived ;echocancel=32 ;or yes ;echocancelwhenbridged=yes channel = 1-15,17-31 The two box works great with my lucent pbx but when i connect the two box the span i have the following error: -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- And want sync the span... Only some time the span is syncronized and i made a one call and works but only for one or two call.. Someone can give me some hits... Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zt_pri_error: PRI: XXX Missing mandatory IE 24/Channel Identification XXX
Hi there ! any hint about this error that I got connecting an eads matra pbx to asterisk with a zaptel pri interface ? my cfgs: zaptel.conf loadzone = us defaultzone = us span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] pridialplan=unknown signalling=pri_net switchtype=euroisdn overlapdial=yes group=1 context=primario channel = 1-15 channel = 17-31 -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Message Extension support
Thanks for the quick, if not very detailed answer. Obviously I am interested in this capability, is there some reason we couldn't work on this? I believe SER might support it (it seems to work between FWD clients at least), why not asterisk? What would be required to implement this functionality? Anyone? I've successfully installed Asterisk and have Microsoft's Instant Messenger connecting. We can make VoIP calls between clients without a problem, however we cannot send text instant messages between clients. From what I can tell this should be possible using IETF SIMPLE or RFC 3428 (SIP Message Extension). I can't find any reference to this and asterisk. Is it supported? No. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and Snom200
--Original Message Text--- From: Barry Fawthrop Date: Thu, 25 Mar 2004 11:07:24 -0500 - Original Message - From: Christian Stredicke To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 10:05 AM Subject: RE: [Asterisk-Users] IAX and Snom200 We thought about this option. I guess the IAX2 is not the problem. We believe the real problem will be the user interface. snom would have no problem providing the platform (hardware plus operating system and stuff like audio), but we simply donĀt want to open another development branch (already got enough trouble with SIP.-). I personally think its ok to optimize the SIP interoperability. All that you can do in IAX can also be done in SIP (or am I making a big mistake here?). Christian There is the big difference. in that IAX handles NAT much better, esp. double NAT (security) Im not sure if you work for snom, but I'm willing to help out where I can. Anyone else care to list the differences between SIP and IAX2? If would be great to get a comprehensive list, Mark or the digium guys ??? Barry I agree that NAT traversal is a huge issue in the SOHO/Small office environment. If the Verbiage (sp?) IAX capable phone is a good product then it will become my preference over the Snom 200. I have several Snom 200s and I like them very much. However, the simplicity of IAX2 installation is too good to pass up. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] ...All we are is dust in the wind. - Kansas ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How detect connection setup/teardown with manager interface?
May problem is: I need to know when and between which channels connection is setup and hungup. Is there a way to learn this from manager interface? There are Link/Unlink events but then appear more than once during single connection, ie while calling from IAX to SIP I get: Event: Link Channel1: [EMAIL PROTECTED]:5036]/3 Channel2: SIP/kamyk-9950 Event: Unlink Channel1: [EMAIL PROTECTED]:5036]/3 Channel2: SIP/kamyk-9950 Event: Link Channel1: [EMAIL PROTECTED]:5036]/3 Channel2: SIP/kamyk-9950 before connection actually is made. Any help? Maciej Kaminski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New soundfiles from Allison posted
I've finally uploaded the newest (LARGE) list of sound clips in .gsm format to the bugtracker. Please see http://bugs.digium.com/bug_view_page.php?bug_id=985 for details and a full sound file list (and a tarball of the sounds in gsm format.) AIF soundfiles are available if you really, really want them, but they're huge and I don't feel like putting them in the bugtracker. As a bonus prize, I have included a clip of Allison saying to hear the full lyrics to louie, louie press... and then of course, the appropriate followup sound clip. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] message waiting notification issues
All, I have some odd message waiting issues with a variety of my SIP clients. Each client has an entry like this in sip.conf; [2200] type=friend host=dynamic context=intern username=2200 secret=2200 dtmfmode=rfc2833 mailbox=2200 As you can see, I specify which a mailbox. This works fine on my Grandstream phones and on the Cisco ATA186 but not with either my Pulver WiSIP or X-Ten Pro (yes I did register it and no you can't have a copy) softphones. I'm also having problems with DIAX which uses IAX. Ideas? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P to E100P for stress test
Looks like you have to have one side of the direct connection supply a clock source. Try having box 2 source the clock on that span: span=1,1,0,ccs,hdb3,crc4,yellow Also, I've never used the Yellow option, so I don't know how that effects things. But anyway, I've done exactly what you want to do, stress test from one system to the other. Should be no problem.. Regards Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Wednesday, March 24, 2004 6:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE410P to E100P for stress test Dear i have two box and i want made some stress test with one TE410P and a E100P with only one span 1 Server TE410P Span1-- PBX Span2---E100P Box The Box with TE410P is Mandrake 9.2 with P4 HT #zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow span=3,0,0,ccs,hdb3,crc4,yellow span=4,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_cpe context=prepaid immediate=no callerid=asreceived channel = 1-15,17-31,63-77,79-93 ; ,125-139,141-155,187-201,203-217 group = 2 switchtype = euroisdn ;signalling = pri_cpe ;signalling = pri_net context=demo ;immediate=yes channel = 32-46,48-62,94-108,110-124 ; ,156-170,172-186,218-232,234-248 The second Box E100P Mandrake 9.2 P3 800Mhz #zaptel.conf /// Add PRI T100P span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_net context=incoming immediate=yes callerid=asreceived ;echocancel=32 ;or yes ;echocancelwhenbridged=yes channel = 1-15,17-31 The two box works great with my lucent pbx but when i connect the two box the span i have the following error: -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- And want sync the span... Only some time the span is syncronized and i made a one call and works but only for one or two call.. Someone can give me some hits... Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immixtel VOIP Adapters
On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote: My wife has a business as well as I. I had envisioned the three line comming in to * in such a manner that her line would be handled in a different manner than my two. Sure. Just change the context for that Zap channel, and have that context set up in extensions.conf. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P to E100P for stress test
Oh, one more thing: You must use an E1 crossover cable when you directly connect one E1 to another (not using a PBX). You can make one yourself, as follows: TEST CABLE WIRING- It's easiest to cut up a standard ethernet CAT5 cable and rewire the connections. Only 4 wires are needed. Please use an ohm meter to make sure you have the wires right! END A END B 1 4 2 5 4 1 5 2 Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com Looks like you have to have one side of the direct connection supply a clock source. Try having box 2 source the clock on that span: span=1,1,0,ccs,hdb3,crc4,yellow Also, I've never used the Yellow option, so I don't know how that effects things. But anyway, I've done exactly what you want to do, stress test from one system to the other. Should be no problem.. Regards Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Wednesday, March 24, 2004 6:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE410P to E100P for stress test Dear i have two box and i want made some stress test with one TE410P and a E100P with only one span 1 Server TE410P Span1-- PBX Span2---E100P Box The Box with TE410P is Mandrake 9.2 with P4 HT #zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow span=3,0,0,ccs,hdb3,crc4,yellow span=4,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_cpe context=prepaid immediate=no callerid=asreceived channel = 1-15,17-31,63-77,79-93 ; ,125-139,141-155,187-201,203-217 group = 2 switchtype = euroisdn ;signalling = pri_cpe ;signalling = pri_net context=demo ;immediate=yes channel = 32-46,48-62,94-108,110-124 ; ,156-170,172-186,218-232,234-248 The second Box E100P Mandrake 9.2 P3 800Mhz #zaptel.conf /// Add PRI T100P span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_net context=incoming immediate=yes callerid=asreceived ;echocancel=32 ;or yes ;echocancelwhenbridged=yes channel = 1-15,17-31 The two box works great with my lucent pbx but when i connect the two box the span i have the following error: -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- And want sync the span... Only some time the span is syncronized and i made a one call and works but only for one or two call.. Someone can give me some hits... Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Semi OT: WiSIP and WEP
Received my Pulver WiSIP phone a couple days ago. Has anyone successfully gotten the phone to work with 128-bit WEP? I've tried entering the key via the keyboard (ugh), turning off WEP then adding the key via the web browser (minor ugh), and all steps in between. The only thing that may be an issue is that my SSID has a space in it Test WAP. When I view it the first time on the phone, it appears correctly. However, the second time, only the first word appears Test. Promising phone if I can ever get it to work on my network. Regards, --- Gavin Adams Promisant (Technology) Ltd. Atlanta, GA smime.p7s Description: S/MIME cryptographic signature
RE: [Asterisk-Users] New soundfiles from Allison posted
John Todd wrote: Please see http://bugs.digium.com/bug_view_page.php?bug_id=985 for details and a full sound file list (and a tarball of the sounds in gsm format.) AIF soundfiles are available if you really, really want them, but they're huge and I don't feel like putting them in the bugtracker. Thank you again for collecting and making all these sounds available for the general public. As a bonus prize, I have included a clip of Allison saying to hear the full lyrics to louie, louie press... and then of course, the appropriate followup sound clip. I thought you were kidding, so I went and looked at the bugtracker. I have absolutely no idea how to respond to that. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 as an IETF Standard?
It looks like we have to create an Internet Draft which is assigned to the relevant working group for revision, questions, comments, more revision, then it may or may not become an RFC. Unfortunately, the IPTel working group appears to be made up of people who are heavily invested in SIP. That's why I suggested an Informational RFC - that is a quicker process that SUN used for NFS. You can publish a spec saying this is how we do it, so you all now. That would be a very good start. I don't think Mark or anyone else have time or interest to fight any protocol wars at this time. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Message Extension support
Hal A. Lightwood wrote: Thanks for the quick, if not very detailed answer. Obviously I am interested in this capability, is there some reason we couldn't work on this? I believe SER might support it (it seems to work between FWD clients at least), why not asterisk? What would be required to implement this functionality? Anyone? Ok, The medium size answer: See http://bugs.digium.com/bug_view_page.php?bug_id=134 * We support some publish/subscribe events implemented in some phones. Snom 200 is an example. I want this tested more thorougly to be able to implement this in a better way. * We support in-dialogue messaging * We do not support Microsoft un-documented un-standardized extensions * We do not support out-of dialogue messaging, you need a voice call According to Mark, this last bullet is because of the current Asterisk architecture. Some of us is discussing ways to solve this without breaking the architecture, but we haven't reached a conclusion on how to make it happen. Yes, using a SIP proxy in front of Asterisk will give you better support for SIP extensions that doesn't involve a call. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immixtel VOIP Adapters
On Thu, 25 Mar 2004 13:51:15 -0500, Tim Sailer wrote: On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote: My wife has a business as well as I. I had envisioned the three line comming in to * in such a manner that her line would be handled in a different manner than my two. Sure. Just change the context for that Zap channel, and have that context set up in extensions.conf. Tim Tim, I understand that as it relates to my present installation with 3 X101p cards in my server. I was wondering if the 4 port FXO adapters from Welltech/Clipcomm/Mediatrix/add-your-favorite-here would work the same way. They're not Zap channels...I think? Do they present themselves as 4 separate ports to *? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] ...I believe in love, its all we've got. - Elton John ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SoftFAX/spandsp
Title: RE: [Asterisk-Users] SoftFAX/spandsp Hi, This is to confirm that with spandsp-0.0.1h Dialogic VFX/40ESC faxing started working, a great deal for us! Thank you, Steve. Will test more ... There is a downside though - looks like this release causes page cutoff. We've had it before - 2 times out of 10. Now it is the case every time with all the machines we've tried. I'm attaching 4 sessions - 2 from J2 and 2 more from the fax machines we have in the office. One page out of 3 is corrupted. Thank you. Alex Zarubin Webley Systems -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED]] Sent: Thursday, March 25, 2004 9:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SoftFAX/spandsp Hi all, My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This version has a number of changes in the way the V.29 modem works. It also has some missing functionality in the T.30 implementation filled in - it was not handling EOM messages. The previous version failed for several reasons with a Dialogic VFX/40ESC. This version succeeds, although it still seems to get a few bit errors, giving some flaws on the received image. I do not see these errors with the other FAX machines I have tried. It seems like a fairly big improvement though, and work will continue to make it better. app_rxfax.c and app_txfax.c have gained a new feature. Previously they always started in answering party mode. Now this is the default behaviour, but something like: exten = 5678,1,txfax(/tmp/testfax.tif|caller) will make them start in calling party mode. So far, these two apps have been little more that testbeds for spandsp. It seems some people are trying to use them for real work, so it seems like they should be gaining more features. The caller mode option was asked for. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 6018.txt.gz Description: Binary data 6020.txt.gz Description: Binary data J2-Fine.txt.gz Description: Binary data J2-regular.txt.gz Description: Binary data
[Asterisk-Users] Voicemail + SIP Message header
I am trying to use Asterisk as a pure voicemail system and have the following setup: I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the To: field, but the INVITE is still being sent to the number asterisk is configured for. Is there any way that I can configure asterisk to read the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number? Thanks Deepak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie and Meetme configuration problem
Hi guys, I am a newbie and having problem to enter a conference room. Here is an extract of my config files: # extensions.conf ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; exten = 8600,1,Meetme,1234 # meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 When I dial 8600 from my SIP soft phone, I get the following error: *CLI Mar 25 20:11:23 WARNING[1184099120]: pbx.c:1179 pbx_extension_helper: No application 'Meetme' for extension (default, 8600, 1) == Spawn extension (default, 8600, 1) exited non-zero on 'SIP/franck-6969' I had a look on the mailing list archive but did not find anything regarding this problem. Thanks in advance for your help Franck
Re: [Asterisk-Users] IAX and Snom200
NAT traversal is a huge issue I agree with Michael and Brian what with the latest viruses etc... security is and will be more and more of an important issue, many SOHO and small corps. Often don't have the know how or finanical backing to implement standard/conventional security and internet access. Thus NAT is so popular. I also agree, that the Verbiage, phone appears a very good product and once out and I have one to test, would look at moving that way too, pressured by the IAX NAT issue. Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail + SIP Message header
Lal, Deepak (Contractor) wrote: I am trying to use Asterisk as a pure voicemail system and have the following setup: I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the To: field, but the INVITE is still being sent to the number asterisk is configured for. Is there any way that I can configure asterisk to read the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number? So all INVITES go to the same URI, regardless of the called number? Is it impossible to change that? If it is, one could implement a SIPTO variable, but I can't see a general need for that. Already have a SIPFROM variable in chan_sip2.c (hint,hint). /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie and Meetme configuration problem
Try doing an answer first: exten = 8600,1,Answer exten = 8600,2,Meetme,1234 Might also be worth doing a Meetme(1234) instead of Meetme,1234. I believe both should work, but.. -Original Message- From: Mailling LIst [mailto:[EMAIL PROTECTED] Sent: Thursday, March 25, 2004 3:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie and Meetme configuration problem Hi guys, I am a newbie and having problem to enter a conference room. Here is an extract of my config files: # extensions.conf ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; exten = 8600,1,Meetme,1234 # meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 When I dial 8600 from my SIP soft phone, I get the following error: *CLI Mar 25 20:11:23 WARNING[1184099120]: pbx.c:1179 pbx_extension_helper: No application 'Meetme' for extension (default, 8600, 1) == Spawn extension (default, 8600, 1) exited non-zero on 'SIP/franck-6969' I had a look on the mailing list archive but did not find anything regarding this problem. Thanks in advance for your help Franck ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie and Meetme configuration problem
Mailling LIst wrote: Hi guys, I am a newbie and having problem to enter a conference room. Here is an extract of my config files: I had a look on the mailing list archive but did not find anything regarding this problem. Thanks in advance for your help This is really a FAQ. You need a Zaptel Timer. Check the Wiki, page Asterisk timer I believe. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Semi OT: WiSIP and WEP
Hi Gavin, Works OK with my 128-Bit WAP. Remove the Space or put in an underscore and try again. Regards Dave -Original Message- Gavin Adams wrote: - Received my Pulver WiSIP phone a couple days ago. Has anyone successfully gotten the phone to work with 128-bit WEP? I've tried entering the key via the keyboard (ugh), turning off WEP then adding the key via the web browser (minor ugh), and all steps in between. The only thing that may be an issue is that my SSID has a space in it Test WAP. When I view it the first time on the phone, it appears correctly. However, the second time, only the first word appears Test. Promising phone if I can ever get it to work on my network. Regards, --- Gavin Adams Promisant (Technology) Ltd. Atlanta, GA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail + SIP Message header
The ${RDNIS} variable in the dialplan would contain that information.
${RDNIS} for SIP is in CVS HEAD. A patch for 0.7.2 is at
http://www.fnords.org/~eric/asterisk/downloads/
On Thu, 2004-03-25 at 14:07, Lal, Deepak (Contractor) wrote:
I am trying to use Asterisk as a pure voicemail system and have the following
setup:
I have the * setup as a SIP peer to a softswitch. When someone calls a number on
the softswitch and no one picks up the phone, the softswitch forwards the call
to the * using SIP. The message header of the SIP INVITE has the number
originally called in the To: field, but the INVITE is still being sent to the
number asterisk is configured for.
Is there any way that I can configure asterisk to read the To: field in the
message header of the SIP INVITE and then go to the mailbox of the corresponding
number?
Thanks
Deepak
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--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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RE: [Asterisk-Users] Semi OT: WiSIP and WEP
A quick search of Yahoo found quite a few reports of issues in various devices with spaces in the SSID. Seems a lot of implementations fail to properly handle the space. Definitely sounds like a WiSIP issue, but might be worth removing the space from your SSID if at all convenient Sean -Original Message- From: David J Carter [mailto:[EMAIL PROTECTED] Sent: Thursday, March 25, 2004 3:53 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Semi OT: WiSIP and WEP Hi Gavin, Works OK with my 128-Bit WAP. Remove the Space or put in an underscore and try again. Regards Dave -Original Message- Gavin Adams wrote: - Received my Pulver WiSIP phone a couple days ago. Has anyone successfully gotten the phone to work with 128-bit WEP? I've tried entering the key via the keyboard (ugh), turning off WEP then adding the key via the web browser (minor ugh), and all steps in between. The only thing that may be an issue is that my SSID has a space in it Test WAP. When I view it the first time on the phone, it appears correctly. However, the second time, only the first word appears Test. Promising phone if I can ever get it to work on my network. Regards, --- Gavin Adams Promisant (Technology) Ltd. Atlanta, GA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Dialing in from the pstn to sip phones (x-lite softphone on winders and a grandstream handytone-286 ata), I hear the sip phone ring a few times, I ran into the same thing with Cisco 7960. Looks like the logic in the sip channel has changed recently. Add a ,r to the end of your Dial statements in extensions.conf and the issue should go away. Does anyone know if this was done intentionally? I don't want to open a bug for something that's really a feature, but i simply can't think of any reason someone want's to update their whole extensions.conf. Can someone tell me what i have to change in the source to get the old (correct :-) way ...? Regards, Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 International Termination
1. Aastra 390 to Digium tdm. 2. E1 line. Redirected DID number to USA. tried with several phones - simemens s45 gsm, panasonic, ge. Everything works fine, but dtnf relaying is broken. Stephen Karrington wrote: Thanks for the feedback. What kind of phone are you using? Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Thursday, March 25, 2004 3:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 International Termination Tested from Bulgaria. The quality is great, even that the ping from here is 170ms. Some troubles with dtmf sending. Stephen Karrington wrote: Hello Everyone, We are about to launch our International IAX2 worldwide termination service from any IAX2 softphone. We would like people to make FREE calls to the USA or Canada so we can check the stability of our platform. We are allowing everyone to call the USA for free RIGHT NOW! You can make calls to any land line phone or mobile phone in the USA and Canada! The string to dial is: [EMAIL PROTECTED]/01510111 This is an example of calling a USA based number. If you want to call a San Francisco number like 1-510-333- then the string to dial is: [EMAIL PROTECTED]/01510333 We are doing this to test a few things and would like your feedback on the following: 1. Call quality. 2. Server loading. We are wondering how many simultaneous calls we can get on this server before it hits too high a load and affects the call quality. Please send any feedback on the call quality and your experience to support AT diamondcard.us. This server is located in the East coast of the USA. All users who are within that vicinity or even on the West Coast should experience very good call quality. Callers from other parts of the world will experience lesser quality depending on their location and how good their internet connection is. We will be implementing servers in Central Europe shortly for European callers to use our service. There might be some downtiime if we have to reconfigure the server to handle issues that arise when people start calling. Thanks for your feedback and have fun making calls to the USA! Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Semi OT: WiSIP and WEP
Will do guys. It didn't even occur to me until I was heading into the office. WiSIP + beer == FATAL_USER_ERROR! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Thursday, March 25, 2004 3:53 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Semi OT: WiSIP and WEP A quick search of Yahoo found quite a few reports of issues in various devices with spaces in the SSID. Seems a lot of implementations fail to properly handle the space. Definitely sounds like a WiSIP issue, but might be worth removing the space from your SSID if at all convenient Sean -Original Message- From: David J Carter [mailto:[EMAIL PROTECTED] Sent: Thursday, March 25, 2004 3:53 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Semi OT: WiSIP and WEP Hi Gavin, Works OK with my 128-Bit WAP. Remove the Space or put in an underscore and try again. Regards Dave -Original Message- Gavin Adams wrote: - Received my Pulver WiSIP phone a couple days ago. Has anyone successfully gotten the phone to work with 128-bit WEP? I've tried entering the key via the keyboard (ugh), turning off WEP then adding the key via the web browser (minor ugh), and all steps in between. The only thing that may be an issue is that my SSID has a space in it Test WAP. When I view it the first time on the phone, it appears correctly. However, the second time, only the first word appears Test. Promising phone if I can ever get it to work on my network. Regards, --- Gavin Adams Promisant (Technology) Ltd. Atlanta, GA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] Immixtel VOIP Adapters
On Thu, Mar 25, 2004 at 01:58:54PM -0600, Michael Graves wrote: On Thu, 25 Mar 2004 13:51:15 -0500, Tim Sailer wrote: On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote: My wife has a business as well as I. I had envisioned the three line comming in to * in such a manner that her line would be handled in a different manner than my two. Sure. Just change the context for that Zap channel, and have that context set up in extensions.conf. Tim Tim, I understand that as it relates to my present installation with 3 X101p cards in my server. I was wondering if the 4 port FXO adapters from Welltech/Clipcomm/Mediatrix/add-your-favorite-here would work the same way. They're not Zap channels...I think? Do they present themselves as 4 separate ports to *? What does ztcfg -v show you? How many channels? You should see info in 'dmesg' also. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_sccp and lamda-solutions
I was following the development of chan_sccp on the Lambda website, but sometime last week all of the links went dead, bugs, cvs, etc. Did the development move? Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreeBSD Segmentation Fault on start up
To all; I've got two installations of asterisk. The last one (installed a few days ago) is from the FreeBSD ports, and many thanks, because it compiled BEAUTIFULLY! However, I can't run it. Everytime I start asterisk, I get a segmentation fault. asterisk -c reveals : [...snip...] [codec_gsm.so] = (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 5 [codec_mp3_d.so] = (MP3/PCM16 (signed linear) Translator (Decoder only)) Segmentation fault (core dumped) So, I check the core dump to see what I can find, and get : Reading symbols from /usr/local/lib/asterisk/modules/codec_gsm.so... (no debugging symbols found)...done. Loaded symbols for /usr/local/lib/asterisk/modules/codec_gsm.so Reading symbols from /usr/local/lib/asterisk/modules/codec_mp3_d.so... (no debugging symbols found)...done. Loaded symbols for /usr/local/lib/asterisk/modules/codec_mp3_d.so Reading symbols from /libexec/ld-elf.so.1...(no debugging symbols found)... done. Loaded symbols for /libexec/ld-elf.so.1 #0 0x2953ff53 in unpack_huff () from /usr/local/lib/asterisk/modules/codec_mp3_d.so (gdb) Would there, by chance, be a missing library or package that I need? Could someone point out a possible solution? (Maybe the port assumed I have an mp3 library installed?) Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX and Snom200
I think Asterisk should have no problem with NAT, even when used with SIP. I mean just listen for the first RTP packet and send the stream where it comes from (thats called symmetrical NAT). I think everybody is doing it like this now and they are selling their stuff for thousands and thousands of dollars. Well we do try to look over our shoulders. There is a lot of tempting stuff out there, and making decisions is difficult. At the moment I think it would be a mistake for us to start another development branch. We simply have too many open issues with SIP already. We hope to have a great phone (some day.-) that fits Asterisk pretty good although its just using SIP Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Mulligan Sent: Thursday, March 25, 2004 6:41 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX and Snom200 Certainly there is the NAT issue and this should not be underestimated. Also IAX allows optimisation of existing bandwidth between Asterisk servers. The SNOM guys should look over their shoulders at Verbiage who are bringing an IAX phone to market. I suspect it will have a lot of interest amongst this community. Brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Barry Fawthrop Sent: 25 March 2004 16:07 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX and Snom200 - Original Message - From: Christian Stredicke To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 10:05 AM Subject: RE: [Asterisk-Users] IAX and Snom200 We thought about this option. I guess the IAX2 is not the problem. We believe the real problem will be the user interface. snom would have no problem providing the platform (hardware plus operating system and stuff like audio), but we simply dont want to open another development branch (already got enough trouble with SIP.-). I personally think its ok to optimize the SIP interoperability. All that you can do in IAX can also be done in SIP (or am I making a big mistake here?). Christian There is the big difference. in that IAX handles NAT much better, esp. double NAT (security) I'm not sure if you work for snom, but I'm willing to help out where I can. Anyone else care to list the differences between SIP and IAX2? If would be great to get a comprehensive list, Mark or the digium guys ??? Barry
Re: [Asterisk-Users] Semi OT: WiSIP and WEP
- Original Message - From: Adams, Gavin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 1:08 PM Subject: [Asterisk-Users] Semi OT: WiSIP and WEP The only thing that may be an issue is that my SSID has a space in it Test WAP. When I view it the first time on the phone, it appears correctly. However, the second time, only the first word appears Test. Are you having problems with the sound clipping a few times every second? I'm using 128bit encryption, the phone registers and can place and receive calls just fine. I'm using ulaw and the sound quality just isn't there. Firmware Version: WF.00.0F Anyone have any hints? Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323.conf, is it possible to track..
Hi all, I am able to track incoming h323 calls with phone number by using amaFlags=billing or amaFlags=documentation. But is it possible to tracking the incoming IP at the same time? If I would like to restrict incoming h323 access to certain IP, should it be done on asterisk or oh323 level? Thanks in advance. Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail + SIP Message header
Lal, Deepak (Contractor) [EMAIL PROTECTED] wrote:
I am trying to use Asterisk as a pure voicemail system and have the
following setup: I have the * setup as a SIP peer to a softswitch. When
someone calls a number on the softswitch and no one picks up the phone,
the softswitch forwards the call to the * using SIP. The message header
of the SIP INVITE has the number originally called in the To: field,
but the INVITE is still being sent to the number asterisk is configured
for.
Is there any way that I can configure asterisk to read the To: field in
the message header of the SIP INVITE and then go to the mailbox of the
corresponding number?
It sounds to me as if you're forwarding all VM calls to a single
extension on the Asterisk box, such as 1000, and are then trying to
work out which mailbox the call should be sent to, with no further IDs
to use as a guide.
If you're only using Asterisk as an answering machine (a bit of a
waste, in my view) then you could forward all calls to individual
extensions on the Asterisk box, so extension 2101 on your switch
would defer to [EMAIL PROTECTED] for VM.
Once you have that, you could capture all incoming calls with a single
context in extensions.conf, such as the following:
[]
exten = _,1,Answer
exten = _,2,Wait(1)
exten = _,3,VoiceMail2(su${EXTEN})
exten = _,4,Hangup
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
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[Asterisk-Users] External and internal SIP do not work together with nat
Here's the main problem I've run into. I'm trying to use FWD with Asterisk, and am behind a nat device (dsl modem with nat built-in, no way to bind the IP directly to a server/PC). I also have a SIP gateway, a Welltech 3502 (it goes by many other names, always see it with the 3502 model number). I am unable to get Asterisk to work with both FWD and the 3502 at the same time. It will work perfectly with one or the other, just not both. Since I'm using NAT, in my sip.conf I have to specify the external IP to get FWD to work. I also have a dynamic IP if that matters, but I found that using a domain name in place of an IP works (i.e., I use externip = myserver.gotdns.com and it works fine). When I comment this out, FWD stops working but the 3502 starts working fine. I ran sip debug on the Asterisk console, and it appears that with the externip value set, it's returning that IP to the 3502 instead of the internal IP. If I could get it to return the internal IP for the 3502, and the external IP for FWD, I think it'd work. Below is my sip.conf, with a few minor changes (edited the dynamic dns domain and callerid numbers, and took out actual FWD username/password). This is the current working configuration; I comment out externip and the 3502 gateway works, and if I uncomment it FWD works. For kicks I have set nat=yes and nat=no for both FWD and the gateway ports (had it set to yes for fwd and no for the gateway, then reversed, then both set to yes, and both set to no...no change). I also changed canreinvite to yes for the gateway ports, with no change. ; ; SIP Configuration for Asterisk ; [general] port = 5061 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = mydomain.gotdns.com localnet = 192.168.1.18 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = biz ; Default for incoming calls register = 255:[EMAIL PROTECTED] [fwd] type=friend secret=mypassword username=55 host=fwd.pulver.com dtmfmode=inband context=biz nat=yes canreinvite=no callerid=Business Line (800) 555-1212 [1001] type=friend username=1001 host=dynamic context=main canreinvite=no txgain=3.5 rxgain=2.5 nat=no [1002] type=friend username=1002 host=dynamic context=main canreinvite=no txgain=3.5 rxgain=2.5 nat=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping voice to exceptionally long queue
Hi, We experienced a problem this week on our asterisk box (ast-2) that has a T1 coming in and talks over IAX2 to a second Asterisk box (ast-1). In the current setup we use ast-2 for outgoing phone-calls only, it takes calls (over IAX2) from ast-1 and routes those calls out over the T1. This morning ast-2 would print out the following error message (1124 times total): Mar 25 04:51:07 DEBUG[163851]: Dropping voice to exceptionally long queue on [EMAIL PROTECTED]/16385 After that it would no longer be possible to make outbound phone calls. Ast-1 showed the IAX2 connection to the crashed/hung ast-2 as 'timed out' (instead of 'registered'). In contrast to that 'iax2 registry show' on the hung ast-2 would still show the IAX2 connection as 'registered'. The exact same thing happened 2 days ago where ast-2 printed the above error message about 800 times before hanging. We had to killall -9 asterisk on ast-2 (because 'stop now' would hang indefinitely) and restart it. After that ast-2 would immediately re-register with ast-1 and everything would start working again. Any pointers of what's wrong here? Regards, Markus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 International Termination
Tested from Belgium Very good quality, sometimes breaking up a little. The phone I used is a Snom200 behind *, gsm codec. Ping times are 110 - 115 ms. Did not try dtmf sending. Robert Sprockeels ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SoftFAX/spandsp
Excellent work Steve. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Thursday, March 25, 2004 10:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SoftFAX/spandsp Hi all, My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This version has a number of changes in the way the V.29 modem works. It also has some missing functionality in the T.30 implementation filled in - it was not handling EOM messages. The previous version failed for several reasons with a Dialogic VFX/40ESC. This version succeeds, although it still seems to get a few bit errors, giving some flaws on the received image. I do not see these errors with the other FAX machines I have tried. It seems like a fairly big improvement though, and work will continue to make it better. app_rxfax.c and app_txfax.c have gained a new feature. Previously they always started in answering party mode. Now this is the default behaviour, but something like: exten = 5678,1,txfax(/tmp/testfax.tif|caller) will make them start in calling party mode. So far, these two apps have been little more that testbeds for spandsp. It seems some people are trying to use them for real work, so it seems like they should be gaining more features. The caller mode option was asked for. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 variants and Asterisk
I see that I can purchase G.729 licenses for my Asterisk server, but I have seen that many phones support a G.729 variant like A or B. Are these suppoted by the same G.729 codec in Asterisk? -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 variants and Asterisk
si funciona con el A y B Miguel Cavazos On Thu, 2004-03-25 at 22:47, Carlos Chavez wrote: I see that I can purchase G.729 licenses for my Asterisk server, but I have seen that many phones support a G.729 variant like A or B. Are these suppoted by the same G.729 codec in Asterisk? -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 variants and Asterisk
Carlos Chavez wrote: I see that I can purchase G.729 licenses for my Asterisk server, but I have seen that many phones support a G.729 variant like A or B. Are these suppoted by the same G.729 codec in Asterisk? B is just the fixed point version of A (from memory) - so it works the same as A. A is a reduced complexity version of G.729 - although they both work with each other. A is just slack when looking for the best representation of your voice. FYI, Digium's codec is G.729A, although it makes little difference ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 182 and *
No. The FXO on the 182 is only usable from the box itself. It's for calling local numbers. Erick Weber V. wrote: Hi to everyone: Does someone know if the ATA 182 works OK with asterisk or should I get a HandyTone 486 instade or an ATA 186 and a FXS to FXO converter Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFAX/spandsp
On Thu, 2004-03-25 at 09:33, Steve Underwood wrote: exten = 5678,1,txfax(/tmp/testfax.tif|caller) There are a zillion fax and tiff formats. I'm trying to figure out what output format I should tell GhostScript to use. Any suggestions on which format to try? These are the formats GhostScript can output: faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4 tifflzw tiffpack -- Eric Wieling [EMAIL PROTECTED] BTEL Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error on * startup
When I start or reload * I always get this error (once). Can someone point me in the right direction to fix this. WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (request) Simon - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk
What kind of specs do I need for a asterisk box that will have a pri for pstn and about 65 sip phones I was thinking a Xeon 3.05 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Drop / Call Tranfer - tranfering a call to a different number.
First, I am very new to this software. If I missed a searchable archive, please point me in the right direction. I am wishing to know if Asterisk can be used to do a Call Drop scenario. This is where someone calls, Asterisk answers, ask for the number that the person wishes to dial, gets the PIN, and then completes the call to the number they desired. Once the connection is completed, this software/service is no longer in the call loop. Typically this scenario is used to offer a wider calling area. Called Metro or Extended Metro in our area. There are many people in area that this feature is not available from their phone company, or they don't want to pay much for it, as they don't make many calls. I am certainly willing to provide more information, but I wanted to find out if Asterisk was even something that could do it- or be modified to do so. Thanks, John Chapman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New minor release of Firefly (now with Speex)
I've put up a new dev version of Firefly (http://www.virbiage.com/firefly/download/firefly-dev.exe) Notable Changes: DTMF now works with SIP Speex codec has been added 1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the Hex address - probably stored in event viewer under control panel) Sorry for the delay but I've completely rewritten how contacts work internally (although it looks exactly the same as it did before). This now allows me to do some sexy things with contacts. Stay tuned I'm aiming for a stable release in two weeks so help me find the bugs. Many thanks to thoses who have -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec Voodoo
I have three * servers that all talk to each just fine, and all talk to other * servers (like NuFone, VoicePulse, etc.). I have hard-phones connected to Sipura SPA-2000s on two of the * servers via a local network connection. The third * server only gets connected to remotely, both with IAX and SIP softphones, and with a roaming Sipura with hard-phones. The setup works well. All of the * servers communicate exclusively with GSM between themselves (and also to NuFone and VoicePulse). The quality is pretty good. The local hard phones are using g711 uLaw (since I think that the X100P cards I believe use uLaw by default as well, but I could be way off on that assumption). Codec transcoding from uLaw to GSM seems to work just fine. From a couple of people who post regularly on this list, I have heard that they have great success with iLBC (and some with Speex as well). I think that NuFone prefers iLBC as well, though it works remarkably well for me with GSM. I did some experiments in forcing my * servers to communicate with each other only with iLBC. When I do that, and can see that they are indeed using iLBC, the quality is horrible. There is long stutter, like every sound is being stretched out. I purchased g729 licenses from Digium for all three servers as well. Using g729 on the Sipura devices yielded no better quality than the built-in g726. However, when I made two * servers communicate only with g729, the quality was marginally better than iLBC, and ridiculously worse than GSM. This was surprising to me. All of this is with a very recent cvs checkout of *, done this past Monday the 22nd I believe. Last point is that if I turn jitterbuffer on (with =yes), then I never hear _any sound_ whatsoever, but there are _no errors_ on either side of the channel. I can see on the CLI that voicemail prompts are being played (for example), but I can't hear anything on either side. Turning jitterbuffer=no immediately restores sound, but the quality only sounds good with GSM. What I don't understand is how some people have success with iLBC, and I don't. I also noticed one or two posts from people that claim that GSM isn't working for them, yet it works really well for me. Are there any settings that I am unaware of (other than the standard allow/disallow directives) that I should be tweaking to make these other codecs work as I understand they should? P.S. One last piece of voodoo, just if anyone knows the answer to this. On occasion, I use DIAX to connect to the remote * server. It works very well, and is the best of the IAX softphones (IMHO). Yesterday, it was working just fine. Today, from a different location (both yesterday and today behind NAT, just from different networks), it connects fine, but I have zero sounds and zero errors. There were _no_ changes to the server or the software setup in between. In the past, I have had trouble using X-Lite to this particular * server. Today, when DIAX wasn't working (neither was iaxcomm, it's not a specific DIAX problem), I tried X-Lite again, and it worked flawlessly... The last bit of info on this is that one of the other * servers is on the same lan as the DIAX client, but on different machines. Both are coming from the same NAT router though. The * machine is in the DMZ, so all packets that are sent to the public side are routed directly to *, and that part works perfectly. I don't know if DIAX is clashing with * packets, but I know this has worked in the past (though it's been 2 weeks since I've tried, and I did cvs up the * server since it last worked...). Thanks in advance to any brave soul who tackles some or all of these questions/issues! :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New minor release of Firefly (now with Speex)
When you use firefly in SIP mode it does not un-register with * on exiting the software Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: Friday, 26 March 2004 11:48 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New minor release of Firefly (now with Speex) I've put up a new dev version of Firefly (http://www.virbiage.com/firefly/download/firefly-dev.exe) Notable Changes: DTMF now works with SIP Speex codec has been added 1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the Hex address - probably stored in event viewer under control panel) Sorry for the delay but I've completely rewritten how contacts work internally (although it looks exactly the same as it did before). This now allows me to do some sexy things with contacts. Stay tuned I'm aiming for a stable release in two weeks so help me find the bugs. Many thanks to thoses who have -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoftFAX/spandsp
Hi Eric, I was all day trying and came up with this: gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 \ -dNOPAUSE -sOutputFile=$TIFFILE -- $PSFILE I'm using a modified version of salsafax/sambafax to enable a print2fax option for windows/linux clients. You add a printer to cups and share it via Samba. Then, you append a line with the fax number in the file you want to be faxed Fax-Nr 3433 and print it to the network printer from any application. The scripts extracts the number and then generates a call file for asterisk. Some ps files cannot be extracted, so I used an OCR application (gocr) to extract the text, maybe its overkill, but it works most of the time (here we send less than ten faxes a day, so its no problem for us). I will clean up the scripts and post them for others to use. Good luck, On Thu, 2004-03-25 at 21:19, Eric Wieling wrote: On Thu, 2004-03-25 at 09:33, Steve Underwood wrote: exten = 5678,1,txfax(/tmp/testfax.tif|caller) There are a zillion fax and tiff formats. I'm trying to figure out what output format I should tell GhostScript to use. Any suggestions on which format to try? These are the formats GhostScript can output: faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4 tifflzw tiffpack -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX drops calls exactly 5 secs into the call
Hi List, Two boxes A has a PRI B terminates SIP devices A --IAX-- B Both on the same switch, same IP network. Call from PSTN to A gets pushed via IAX to B - Sip device with no problems. Call from Sip device - B via IAX - A - PSTN will drop exactly 5 seconds after the call is answered. I've built with 0.7.2, 1_0_Stable develetc Any clue / hints ?? thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
