Re: [asterisk-users] AGI and some informations
Ok, i understand,
But i don't know how to get the IP Adress when a softphone is registred,
and how to send to this IP adress, and call number to the softphone, for
an incoming call.
best regards,
Olivier S
Anton Frolov a écrit :
it was not a real code, but just a schema.
I can't write a more precise snippet of code, since I'm completely
unaware of your configuration.
But in any case, I would delegate all of the logic to your Ruby script
and keep the minimum in the extensions.conf.
AF.
Olivier Saulnier wrote:
Anton Frolov a écrit :
When registering the softphone:
OK, in which file do i do that??
SoftPhonesDB.insert(olivier, $ip);
Could you explain me what means SoftPhonesDB.insert?? It's not an AGI
commande, how can i use it??
In extensions.conf:
exten = 302,s,agi,script.rb,${EXTEN}
OK, i understand
In script.rb:
$ip = SoftPhonesDB.select(olivier);
Dial(SIP/$ip, $arg);
Hummm...Does Dial working with Ip adress?? Or should i write ine the
database the softphone number too??
Best regards,
Olivier S
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STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
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Re: [asterisk-users] Custom Voicemail Notification Email
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: You can have your own external script to do whatever you want when vm is left from voicemail.conf: ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp M Hi Marnus, externnotify, of course. I always end up spending months away from asterisk so by the time I come back to it, I've forgotten half the stuff. Thanks for the reminder. Now, maybe I'm stupid but how exactly do I get details to it regarding all those VM variables that are inserted when the email is normally sent out from voicemail. You know the VM_NAME, VM_DUR etc etc? I quickly tested this but as per the doco. for it, it automatically passes only 3 variables to the externotify script. Do I go parsing msg.txt file for the rest of the info? I may not have that in the case I'm using RealTime Voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which SIP transport from France and termination services in the Nederlands
Hi, This question is both technical and business related. I've got a prospective customer in France which belongs to Hotel industry. He has a lot of visitors coming from the Nederlands. I'm studying the opportunity to offer phone services to those visitors. The service I'm thinking about is plain local call termination : hotel guests cost effectively call their relatives in their home country (the Nederlands in this case). The setup is : SIP hardphones in France--LAN -- Asterisk SDSL-- www ? or MPLS network ? ---Gateway PSTN - Phone in the Nederlands I'm wondering what kind of service I should buy to get reliable SIP call transportation (from France to the Nederlands) and local termination (in the Nederlands). Maybe I should just use the Internet and shouldn't care too much about call quality. Any suggestion ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Resolved: Re: [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound
There was something screwy going on with kernel vs kernel-devel. So I rolled back to kernel-*-2.6.9-42 rather than kernel-*-2.6.9-42.0.3. Zaptel has now installed successfully. I don't believe this is a problem with 2.6.9-42.0.3 per se. Rather my system had different versions of kernel vs kernel-devel. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail Notification Email
Your script will have to read the extra info from the msg.txt files or it's realtime equivalent. M Now, maybe I'm stupid but how exactly do I get details to it regarding all those VM variables that are inserted when the email is normally sent out from voicemail. You know the VM_NAME, VM_DUR etc etc? I quickly tested this but as per the doco. for it, it automatically passes only 3 variables to the externotify script. Do I go parsing msg.txt file for the rest of the info? I may not have that in the case I'm using RealTime Voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail Notification Email
You could of course edit app_voicemail.c to pass more info...
Round about line 2329:
if (!ast_strlen_zero(externnotify)) {
if (messagecount(ext_context, newvoicemails,
oldvoicemails)) {
ast_log(LOG_ERROR, "Problem in calculating
number of voicemail messages available for extension %s\n", extension);
} else {
snprintf(arguments, sizeof(arguments), "%s
%s %s %d", externnotify, context, extension, newvoicemails);
ast_log(LOG_DEBUG, "Executing %s\n", arguments);
ast_safe_system(arguments);
}
}
M
RR wrote:
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You can have your own external script to do
whatever you want when vm is
left
from voicemail.conf:
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
M
Hi Marnus,
externnotify, of course. I always end up spending months away from
asterisk so by the time I come back to it, I've forgotten half the
stuff. Thanks for the reminder. Now, maybe I'm stupid but how exactly
do I get details to it regarding all those VM variables that are
inserted when the email is normally sent out from voicemail. You know
the VM_NAME, VM_DUR etc etc? I quickly tested this but as per the
doco. for it, it automatically passes only 3 variables to the
externotify script. Do I go parsing msg.txt file for the rest of
the info? I may not have that in the case I'm using RealTime
Voicemail.
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[asterisk-users] Play an announcement while receiving DTMF?
I have a customer who wants their Asterisk system to play an announcement to the caller while a DTMF string is being sent from the caller's phone. As far as I am aware, this isn't possible: it's only possible to detect DTMF if the first incoming digit interrupts the announcement. Is that correct? Is there any way to achieve what the customer is asking for, or is it an unrealistic request? Thanks in advance for any advice. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iptables example
I use BFD on several of my servers. Works great. http://www.rfxnetworks.com/bfd.php - Original Message - From: Jeronimo Romero To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 28, 2006 11:54 PM Subject: [asterisk-users] iptables example Hey everyone. I recenty installed a server at a datacenter offsite and the thing is getting hammered with invalid ssh logins so I decided to use some iptables. I included my ruleset here. I was wondering if I could get some feedback based on my ruleset from those of you using iptables in production systems. It seems to be working but some critique would be appreciated. Thanks #!/bin/sh # My system IP/set ip address of server SERVER_IP=x.x.x.x # Flushing all rules iptables -F iptables -X # Setting default filter policy iptables -P INPUT DROP iptables -P OUTPUT DROP iptables -P FORWARD DROP # Allow unlimited traffic on loopback iptables -A INPUT -i lo -j ACCEPT iptables -A OUTPUT -o lo -j ACCEPT # Allow incoming ssh only from secure hosts iptables -A INPUT -p tcp -s x.x.x.x -d $SERVER_IP --sport 513:65535 --dport 22 -m state --state NEW,ESTABLISHED -j ACCEPT iptables -A INPUT -p tcp -s x.x.x.x -d $SERVER_IP --sport 513:65535 --dport 22 -m state --state NEW,ESTABLISHED -j ACCEPT #Allow http Asterisk Related Traffic iptables -A INPUT -p tcp -i eth0 --dport 80 -m state --state NEW -j ACCEPT # SIP on UDP iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT # IAX2- iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX - iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT iptables -A INPUT -j DROP iptables -A OUTPUT -j ACCEPT -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Something similar or better than HUD Pro?
Is there something similar or better than HUD pro out there for asterisk PBX. HUD pro is wonderful thing, but they require complete Fonality product to be purchased first, and don't sell it as a stand alone product. If someone is not interested in Fonality product but is ready to purchase some good interface like HUD, what are the options? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Port 5060
Its easy. In sip.conf, under each phone/DID setting, define a different port. e.g. If 8789092323 is assigned to a person in country abc where SIP is blocked, in your sip.conf, under [8789092323] add port=12000 or whatever you like. This will override the default port=5060 setting in sip.conf. I've tried this and it worked for me. But now on the other end the users will also need to change their ports. You'll have to tell them how to change it in their soft/hard phones. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to dial apps always show from asterisk
I have seen the answer to this question previously, perhaps I am just not asking the question correctly. For manager-based apps that do not explicitly set a callerid is there anyway to overide the system default of asterisk On 11/28/06, Tim Panton [EMAIL PROTECTED] wrote: On 28 Nov 2006, at 03:01, Eric Bishop wrote: I am trying to do it with FOP and Calling Circles. Both have closed code. Anyway to do it from Asterisk? You could use the 'Local' channel as the argument to the originate command and then set it in the dialplan. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendmail or postfix?
For voicemail to email solution, just wanted to ask the experts, which one is better and why: sendmail or postfix, or something other. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail Notification Email
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You could of course edit app_voicemail.c to pass more info...
Round about line 2329:
if (!ast_strlen_zero(externnotify)) {
if (messagecount(ext_context, newvoicemails,
oldvoicemails)) {
ast_log(LOG_ERROR, Problem in calculating number
of voicemail messages available for extension %s\n, extension);
} else {
snprintf(arguments, sizeof(arguments), %s %s %s
%d, externnotify, context, extension, newvoicemails);
ast_log(LOG_DEBUG, Executing %s\n, arguments);
ast_safe_system(arguments);
}
}
M
Right, I was looking at the sending email part to instead of sending
out the email, write a file with the relevant info and then I use
externnotify to pick up that file stick it into a template and send it
out.
Also, note changing the Content Type: text/html and recompiling has
allowed me to send and display emails as HTMLS. I'll just frigging
create an entire HTML page as one LONG string with pics and stuff and
give that a GO :)
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[asterisk-users] Re: SIP group management
JO == J Oquendo [EMAIL PROTECTED] writes: JO One thing I noticed about Asterisk and group rings is, if a phone JO is not registered but in the group, sometimes it won't ring. What did you expect? If it isn't registered, Asterisk doesn't know how to reach it, and therefore it doesn't ring. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendmail or postfix?
On Wed, Nov 29, 2006 at 04:48:37AM -0500, Zeeshan Zakaria wrote: For voicemail to email solution, just wanted to ask the experts, which one is better and why: sendmail or postfix, or something other. As far as carrying voicemail messages and delivering them on a system that does not have a high load of messages, they are both quite good. Note that postfix (as well as exim and others) provides /usr/sbin/sendmail binary that Asterisk uses to send mail. I'd start with whatever is the default with the distro, or whatever you're more comfortable with setting up. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP transport from France and termination services in the Nederlands
Hi, I would advice you to buy some minutes at a (or several for redundancy) SIP provider located in France, so you don't need anything in Netherlands... As you have SDSL, connexion between you and the provider should be really good. If you don't need incoming calls from Netherlands through SIP, it's even easier... BR Le Mercredi 29 Novembre 2006 09:35, Olivier a écrit : Hi, This question is both technical and business related. I've got a prospective customer in France which belongs to Hotel industry. He has a lot of visitors coming from the Nederlands. I'm studying the opportunity to offer phone services to those visitors. The service I'm thinking about is plain local call termination : hotel guests cost effectively call their relatives in their home country (the Nederlands in this case). The setup is : SIP hardphones in France--LAN -- Asterisk SDSL-- www ? or MPLS network ? ---Gateway PSTN - Phone in the Nederlands I'm wondering what kind of service I should buy to get reliable SIP call transportation (from France to the Nederlands) and local termination (in the Nederlands). Maybe I should just use the Internet and shouldn't care too much about call quality. Any suggestion ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_misdn on a junghanns card
Hello, I am trying to use chan_misdn on a junghanns QuadBRI card. Using the latest install-misdn-mqueue from beronet, all installation went well apparently. However when I try to load the card it is not recognized: # modprobe hfcmulti type=0x04 protocol=0x12,0x12,0x2,0x2 layermask=0x3,0x3,0xf,0xf Loading only hfcmulti - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x4 protocol=0x12,0x12,0x2,0x2 layermask=0x3,0x3,0xf,0xf poll=64 debug=0 Nov 29 11:42:45 pyrrhus kernel: 0 devices registered Trying the same thing on a hfcpci card works and I can receive call with chan_misdn. Is there something specific to junghanns cards? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring an asterisk server during off hours
Hi, During off hours, a server of mine simply forward incoming calls to an outside number, so that no user is locally available to report or notify downtimes. As availability is here a major requirement, I'm looking for a cost effective and reliable way to monitor this server. Should I simply call every 10 minutes, a dedicated extension to check PSTN lines and server availability or is there a smarter way to do it ? Setup: Nagios - Monitoring Asterisk -- PSTN -- Monitored Asterisk - VPN access --www --- Back to Monitoring Asterisk With this a single check would test PSTN lines, asterisk server and VPN access availability. I don't think it should be very difficult to trigger a call from Nagios. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: chan_misdn on a junghanns card
On Wed, Nov 29, 2006 at 11:45:50AM +0100, Louis-David Mitterrand wrote: Hello, I am trying to use chan_misdn on a junghanns QuadBRI card. Using the latest install-misdn-mqueue from beronet, all installation went well apparently. However when I try to load the card it is not recognized: This card is a new-style QuadBRI v 2.0 (with hardware watchdog according to KP Junghanns). Apparently this new card is not recognized by mISDN drivers. I tried using an old QuadBRI and the modules load fine. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring an asterisk server during off hours
Somewhere did I see a test script. I will see if I can find it once more. With that information should you be able to write a simple script that monitor the server and then will notify you if the server stop responding. PING wold maybe also be a help. //Mattias On 29/11/06, Olivier [EMAIL PROTECTED] wrote: Hi, During off hours, a server of mine simply forward incoming calls to an outside number, so that no user is locally available to report or notify downtimes. As availability is here a major requirement, I'm looking for a cost effective and reliable way to monitor this server. Should I simply call every 10 minutes, a dedicated extension to check PSTN lines and server availability or is there a smarter way to do it ? Setup: Nagios - Monitoring Asterisk -- PSTN -- Monitored Asterisk - VPN access --www --- Back to Monitoring Asterisk With this a single check would test PSTN lines, asterisk server and VPN access availability. I don't think it should be very difficult to trigger a call from Nagios. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens Gigaset C450 IP vs S450 IP
I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? -- Eugen* Leitl a href=http://leitl.org;leitl/a http://leitl.org __ ICBM: 48.07100, 11.36820http://www.ativel.com 8B29F6BE: 099D 78BA 2FD3 B014 B08A 7779 75B0 2443 8B29 F6BE signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 11:26 AM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP Port 5060 (Tom Lynn)
your client have to solicitate at the ISP the unblock of this port,i live in argentina, its a normal that happens, your client have to solicitate tue unblock of that port. Saludos Leonardo FREE pop-up blocking with the new MSN Toolbar MSN Toolbar Get it now! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten = _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten = _.,2,LookupCIDName
;exten = _NXXNXX,3,Dial(sip/sammy,30,r)
;exten = h,1,HangUp()
;exten = s,1,Dial(SIP/timothy)
;exten = s,2,Hangup()
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
As you can see I tried a few things, but none of them work.
Does anybody know how to solve this ?
Thnx.
___
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Re: [asterisk-users] IAX access to FWD broken?
I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. Timothy Parez wrote: I've got the same problem here. It can't register anymore -- timeout Brian Capouch schreef: I hadn't used FWD for quite a while. A customer sent me an email last week, Is FWD broken when one tries to use it with IAX? I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 11:26 AM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX access to FWD broken?
I just sent an e-mail to the FWD support address, I'll let you know where that gets me. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. Timothy Parez wrote: I've got the same problem here. It can't register anymore -- timeout Brian Capouch schreef: I hadn't used FWD for quite a while. A customer sent me an email last week, Is FWD broken when one tries to use it with IAX? I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset C450 IP vs S450 IP
Hi Eugen, Eugen Leitl wrote: I've just ordered a Siemens Gigaset C450 IP cordless IP/DECT phone, given that it's supported by asterisk http://www.voipuser.org/review_41.html However, I see that a slightly better Gigaset S450 IP is available for only a slight price premium. Are there any user experiences with the S450 IP? Unfortunately I haven't yet had one in my hands, but from the feature-list it seems a bit more value for money compared to the C450. Especially being able to handle 4 SIP accounts/lines at one provider, being able to add more handsets to the basestation etc. would be of value for SOHO use. I did notice the C450 is unable to use the flash-key for call transfer functionality with SIP accounts, which is a bit of a shame. I'm not sure if that will be supported with the S450. If anyone can shed a light on that I'd be interested as well :-) Florian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and some informations
Hello, I'll try to explain better what i want to do: 1 - i've developped a softphone, with iaxclient.dll, in Windev Langage (French langage - PC SOFT). This DLL doesn't work well with Windev, some of pieces are ok. the ring, is not OK!! 2 - For detect the ring, i have make a listen server on the softphone. On the Asterisk server, i've make a Ruby Script which give the information as the RING must bell, and when it must stop. 3 - I modify the extensions.conf file fir call the Ruby script. but it's not a good work, because: a) I must specify for each softphone a new context, where i call the script with the IP address. b) i must give at softphone the phone nimber incoming, for external calls Do you have any idea for process that?? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset C450 IP vs S450 IP
On Wed, Nov 29, 2006 at 01:49:28PM +0100, Florian Overkamp wrote: Unfortunately I haven't yet had one in my hands, but from the feature-list it seems a bit more value for money compared to the C450. Especially being able to handle 4 SIP accounts/lines at one provider, being able to add more handsets to the basestation etc. would be of value for SOHO use. I've done a more thorough websearch, and came up with http://www.ip-phone-forum.de/showthread.php?t=120442 /kraut There's also a post on some swedish discussion board mentioning asterisk, which I'm unfortunately unable to read. I did notice the C450 is unable to use the flash-key for call transfer functionality with SIP accounts, which is a bit of a shame. I'm not sure if that will be supported with the S450. If anyone can shed a light on that I'd be interested as well :-) -- Eugen* Leitl a href=http://leitl.org;leitl/a http://leitl.org __ ICBM: 48.07100, 11.36820http://www.ativel.com 8B29F6BE: 099D 78BA 2FD3 B014 B08A 7779 75B0 2443 8B29 F6BE signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 12:41 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 11:26 AM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Desktop application for zap/agent call control
I am looking for a desktop control panel for zap (agent proxies). Does any one know of an application that is similiar to a softphone but controls zap/agents interfaces. I am looking for phone book, transfer, and possibly presence control. And it must be standalone, unlike HUD pro and hudLite. Jordan Novak Senior Telecommunications Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] keep line on hook
Hi all, is there a way I can put a line on hook ? I'd like to keep the line busy on demand (es. dialing an extension will put on hook line n.1) so the caller receives busy tone directly from PSTN and not from asterisk. Thanks. marco ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Hi believe that you nead a standalone image. Would you consider use SIP image, that could be possible to find on the net. //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 12:41 PM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - *From:* Mattias Andersson [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, November 29, 2006 11:26 AM *Subject:* Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - *From:* Admin @ TheAdmiralNelson.Com [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Thursday, November 23, 2006 6:54 PM *Subject:* [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten = _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten = _.,2,LookupCIDName
;exten = _NXXNXX,3,Dial(sip/sammy,30,r)
;exten = h,1,HangUp()
;exten = s,1,Dial(SIP/timothy)
;exten = s,2,Hangup()
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
I had a similar issue. If you turn on the chan_misdn debug messages than
you can see what chan_misdn sees as the incoming number. My problem was
that my dialplan had for example 881234567 while chan_misdn was seeing
0881234567 so there was no match. A quick change from 881234567 to
0881234567 in my dialplan fixed it.
Regards,
Patrick
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[asterisk-users] Re: IAX access to FWD broken?
Just as an it works for me, I created a FWD account a couple of weeks ago, which seems to be working fine. I am able to receive calls over IAX2 via my IpKall number. Jim Timothy Parez wrote: I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)
Sorry to bother you all with what is probably a simple question.
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
My script seems to execute ok, and I can see asterisk playing the sounds but
my phone goes from ringing to busy, and I don't hear the phontics.
Below are the relevant bit from my PHP, Console, and extensions.conf.
I would be most grateful if someone could show me the way.
Thanks in advance:
Chris
Asterisk ver: 1.2.10
PHP:
#!/usr/local/php/bin/php -q
?php
$stdin = fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/var/log/asterisk/my_agi.log', 'w');
while (!feof($stdin)) {
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if (($temp == ) || ($temp == \n)) {
break;
}
}
fputs($stdout,SAY PHONETIC \abc\ \#\ \n);
fflush($stdout);
$msg = fgets($stdin,1024);
fputs($stdlog,$msg . \n);
?
Extensions.conf:
exten = 4343,1,Answer
exten = 4343,2,AGI(example.php)
exten = 4343,3,Busy
AGI Debug:
AGI Rx SAY PHONETIC abc #
-- Playing 'phonetic/a_p' (language 'en')
-- Playing 'phonetic/b_p' (language 'en')
-- Playing 'phonetic/c_p' (language 'en')
-- AGI Script example.php completed, returning 0
-- Executing Busy(SIP/4321-081b9498, ) in new stack
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Re: [asterisk-users] Re: IAX access to FWD broken?
FWD works fine for me. I just set up a trunk in asterisk. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Jim Lawson wrote: Just as an it works for me, I created a FWD account a couple of weeks ago, which seems to be working fine. I am able to receive calls over IAX2 via my IpKall number. Jim Timothy Parez wrote: I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0651-2, 11/28/2006 - 11/29/2006 9:27:39 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten = _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten = _.,2,LookupCIDName
;exten = _NXXNXX,3,Dial(sip/sammy,30,r)
;exten = h,1,HangUp()
;exten = s,1,Dial(SIP/timothy)
;exten = s,2,Hangup()
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
I had a similar issue. If you turn on the chan_misdn debug messages than
you can see what chan_misdn sees as the incoming number. My problem was
that my dialplan had for example 881234567 while chan_misdn was seeing
0881234567 so there was no match. A quick change from 881234567 to
0881234567 in my dialplan fixed it.
Regards,
Patrick
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Re: [asterisk-users] Re: IAX access to FWD broken?
I'm travelling today but I was just able to use Firefly to login to FWD via IAX2. I called the echo test with no problems other than the lousy network in this hotel. My Astlinux server Also reports that it's registered with FWD via IAX2. My account is a couple years old. Michael On Wed, 29 Nov 2006 09:49:25 -0500, Al Bochter wrote: FWD works fine for me. I just set up a trunk in asterisk. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Jim Lawson wrote: Just as an it works for me, I created a FWD account a couple of weeks ago, which seems to be working fine. I am able to receive calls over IAX2 via my IpKall number. Jim Timothy Parez wrote: I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0651-2, 11/28/2006 - 11/29/2006 9:27:39 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP upgrade issues
Thanks for all the help guys. I cannot load the new SIP image straight on as the SCCP image is very old. i read the FAQs posted on the lists and it tells me I need to upgrade the SCCP image to at least 7 before I can load the SIP image. This is the problem I am having. I cannot load SIP until I have at least V7 of SCCP. I downloaded the SCCP image but when you run setup it comes back with a registry error making me think it needs a call manager. Has anyone EVER managed to load the SIP image onto a 7970 that had V5 code? Thanks - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 2:15 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi believe that you nead a standalone image. Would you consider use SIP image, that could be possible to find on the net. //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Its not even at the tftp stage. When I run the image file from Chisco and attempt to run setup I get a registry error. I am assuming its because its expecting a call manager. How do I upgrade the firmware? The image I have is only for callmanager cmterm-7970_7971-sccp.7-0-2SR1 Anyone know of a standalone image? - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 12:41 PM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi Paul! I do thing you could use a TFTP bout I have not ben woring with that phone. Could you post your TFTP loog? //Mattias On 29/11/06, Paul A Brown [EMAIL PROTECTED] wrote: Hi Mattias, That is what I did for my 7960 and what I need to do for this. However my problem is when I un tar the cisco file it won't run. I think it needs call manager :-( Paul - Original Message - From: Mattias Andersson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, November 29, 2006 11:26 AM Subject: Re: [asterisk-users] Cisco 7970 SIP upgrade issues Hi! I have only used 7940 and 7905. The 7940 are supporting TFTP and I did use that to upgrade them. I had to do it in 3 steps. First a old SIP firmware. Then an newer firmware and then the on that I am using. //Mattias On 29/11/06, Paul [EMAIL PROTECTED] wrote: Does anyone have any ideas? I am pulling my hair out :-) I changed email address's which is why the names different. Thanks in advance - Original Message - From: Admin @ TheAdmiralNelson.Com To: asterisk-users@lists.digium.com Sent: Thursday, November 23, 2006 6:54 PM Subject: [asterisk-users] Cisco 7970 SIP upgrade issues Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Voicemail, SQL ODBC
Derek Whitten wrote: Norbert Zawodsky wrote: RR wrote: snip Mate, I can't say it with authority but I'm almost certain that the only DB that a specific driver was written for is MySQL. I think if you use res_mysql.o you should be able to talk to mySql directly without needing ODBC. /snip O.k., Nice to hear. But I'm not sure *how* to use res_mysql.o. In other words: I configured cdr_mysql.conf and asterisk happily writes CDRs into my mySQL DB on a different server. But how should I do that regarding voicemail? Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Voicemail This one stores voicemail *config* using RealTime, *NOT* voicemail data! http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage This one is how to store viocemail-data using ODBC. The question of this thread was how to store voicemail data *NOT* using ODBC. Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, SQL ODBC
Noah Miller wrote: Hi Peder - Is the storage of actual voicemail messages in a database still limited to ODBC? If so, why? Yes. Why? Nobody has developed a voicemail solution that directly connects to a *SQL database for message storage. A clear answer :-) Although a sad one :-( Because it means, I have to install all that ODBC stuff (which always lead to trouble in the past). I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a separate SQL box and none of it should affect *. Yes, you can do this. There are other ways to do this without using database storage for voicemail (e.g. you could use normal file-based storage for voicemail and have the webserver connect to the vm storage on the * server via nfs). In my case, that's not possible. Because the box asterisk runs on has very limited memory an no harddisk. I don't know for Peder, but for me the only way seems to be to install ODBC. Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blind transfer # not working for forwarded or picked calls
Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 Kind regards Roger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX access to FWD broken?
Still doesn't work for me. Still get timeout Michael Graves schreef: I'm travelling today but I was just able to use Firefly to login to FWD via IAX2. I called the echo test with no problems other than the lousy network in this hotel. My Astlinux server Also reports that it's registered with FWD via IAX2. My account is a couple years old. Michael On Wed, 29 Nov 2006 09:49:25 -0500, Al Bochter wrote: FWD works fine for me. I just set up a trunk in asterisk. Best regards, Al Bochter Bochter Services _http://www.BochterServices.com/?t=Email_ (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: _http://www.freeworlddialup.com/_ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * _http://www.bochterservices.com/?t=TF(NM)did_ BUY Coins, Silver and Gold _http://www.bochterservices.com/?j=goldt=email_ For new and used security items _http://www.bochterservices.com/?j=storet=email_security_ Jim Lawson wrote: Just as an it works for me, I created a FWD account a couple of weeks ago, which seems to be working fine. I am able to receive calls over IAX2 via my IpKall number. Jim Timothy Parez wrote: I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: _http://lists.digium.com/mailman/listinfo/asterisk-users_ Inbound (clean). Database: 0651-2, 11/28/2006 - 11/29/2006 9:27:39 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: _http://lists.digium.com/mailman/listinfo/asterisk-users_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
Hi,
I did, that was my first try,
but it didn't work.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten = _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten = _.,2,LookupCIDName
;exten = _NXXNXX,3,Dial(sip/sammy,30,r)
;exten = h,1,HangUp()
;exten = s,1,Dial(SIP/timothy)
;exten = s,2,Hangup()
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
I had a similar issue. If you turn on the chan_misdn debug messages than
you can see what chan_misdn sees as the incoming number. My problem was
that my dialplan had for example 881234567 while chan_misdn was seeing
0881234567 so there was no match. A quick change from 881234567 to
0881234567 in my dialplan fixed it.
Regards,
Patrick
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[asterisk-users] Asterisk + Avaya S8700
Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments regarding the matter. The purpose of Asterisk on this matter is to provide outgoing calls from the Avaya through Asterisk, so features such as MWI and stuff are not necessary for me. Thanks, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
I get the following with debug on:
P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none
P[ 3] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] -- Bearer: Speech
P[ 3] -- Codec: Alaw
P[ 0] -- * NEW CHANNEL dad:50556010 oad:497978546
P[ 3] -- CTON: Unknown
P[ 3] EXPORT_PID: pid:10
P[ 3] -- PRES: Restricted (0)
P[ 3] -- SCREEN: Unscreened (0)
Nov 29 16:39:40 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
P[ 3] I SEND:RELEASE oad:0497978546 dad:050556010 pid:10
P[ 3] -- bc_state:BCHAN_CLEANED
P[ 3] -- channel:1 mode:TE cause:16 ocause:1 rad: cad:
P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] I IND :RELEASE_COMPLETE oad: dad: pid:10 state:EXTCANTMATCH
P[ 3] -- channel:0 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0
P[ 3] hangup_chan
P[ 3] - hangup
P[ 3] * IND : HANGUPpid:10 ctx:inisdn dad:050556010 oad:0497978546
State:EXTCANTMATCH
P[ 3] -- l3id:6000b
P[ 3] -- cause:16
P[ 3] -- out_cause:16
P[ 3] -- state:EXTCANTMATCH
P[ 3] Channel: mISDN/3-1 hanguped new state:CLEANING
P[ 3] release_chan: bc with l3id: 6000b
so I change extensions.conf:
exten 50556010,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten 50556010,2,Answer()
exten 50556010,3,Echo()
exten 50556010,4,Hangup()
And the debug message above is what I get
Timothy.
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Re: [asterisk-users] IAX access to FWD broken?
jason wrote: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. Timothy Parez wrote: I've got the same problem here. It can't register anymore -- timeout Brian Capouch schreef: I hadn't used FWD for quite a while. A customer sent me an email last week, Is FWD broken when one tries to use it with IAX? I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have several FWD accounts and they all work fine with IAX... signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
Hi,
Sorry, uncomenting that actually worked.
Now I need to filter on the last two numbers, that shoulnd't be to hard
I guess.
Tim.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten = _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten = _.,2,LookupCIDName
;exten = _NXXNXX,3,Dial(sip/sammy,30,r)
;exten = h,1,HangUp()
;exten = s,1,Dial(SIP/timothy)
;exten = s,2,Hangup()
;exten = _X.,1,Dial(SIP/timothy,30,r)
;exten = _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
I had a similar issue. If you turn on the chan_misdn debug messages than
you can see what chan_misdn sees as the incoming number. My problem was
that my dialplan had for example 881234567 while chan_misdn was seeing
0881234567 so there was no match. A quick change from 881234567 to
0881234567 in my dialplan fixed it.
Regards,
Patrick
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[asterisk-users] b410p hangup detection - Portugal
Hi, I've setup a trixbox 1.2.2 instalation with digium b410p. It uses misdn driver. I'm able to place calls and receive calls using this interface. One thing that happens is when i dial an a number from a sip client to an number that is routed through b410p and the called party rejects the call, asterisk doesn't detect the hangup. The sip client remains in ringing waiting for the call. Does anyone have any info or similar problems? Thanks, -- Nuno Miguel Pais Fernandes [EMAIL PROTECTED] Cisco Certified Network Associate Oracle Certified Professional Eurotux Informática, S.A. [http://eurotux.com] Rua Rosalvo de Almeida, 5. 4710-429 BRAGA PORTUGAL Tel: (+351) 253 257395 - Fax: (+351) 253 257396 pgpQkUzKKwiAx.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX access to FWD broken?
I've got to the point with FWD and IAX that I just connect directly via SIP to IPKall, using my Asterisk box's address as the proxy. It simply works better and eliminates another point of failure at FWD. I also find it helps keep things a little more organized since I can assign my own internal SIP extensions to map to the IPKall number, so I can see what is going where with just a quick glance. Z On 11/29/06, Jim Lawson [EMAIL PROTECTED] wrote: Just as an it works for me, I created a FWD account a couple of weeks ago, which seems to be working fine. I am able to receive calls over IAX2 via my IpKall number. Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3
Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c - app_cepstral.o In file included from /usr/src/asterisk/include/asterisk/linkedlists.h:23, from /usr/src/asterisk/include/asterisk/frame.h:37, from /usr/src/asterisk/include/asterisk/channel.h:110, from app_cepstral.c:33: /usr/src/asterisk/include/asterisk/lock.h: In function `ast_mutex_init': /usr/src/asterisk/include/asterisk/lock.h:513: warning: implicit declaration of function `pthread_mutexattr_settype' /usr/src/asterisk/include/asterisk/lock.h:513: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/src/asterisk/include/asterisk/lock.h:513: error: (Each undeclared identifier is reported only once /usr/src/asterisk/include/asterisk/lock.h:513: error: for each function it appears in.) In file included from /usr/src/asterisk/include/asterisk/cdr.h:48, from /usr/src/asterisk/include/asterisk/channel.h:115, from app_cepstral.c:33: /usr/src/asterisk/include/asterisk/utils.h: In function `_ast_strndup': /usr/src/asterisk/include/asterisk/utils.h:421: warning: implicit declaration of function `strndup' /usr/src/asterisk/include/asterisk/utils.h:421: warning: assignment makes pointer from integer without a cast /usr/src/asterisk/include/asterisk/utils.h: In function `_ast_asprintf': /usr/src/asterisk/include/asterisk/utils.h:446: warning: implicit declaration of function `vasprintf' In file included from app_cepstral.c:36: /opt/swift/include/swift.h: At top level: /opt/swift/include/swift.h:765: warning: function declaration isn't a prototype app_cepstral.c:43: warning: type defaults to `int' in declaration of `STANDARD_LOCAL_USER' app_cepstral.c:43: warning: data definition has no type or storage class app_cepstral.c:44: warning: type defaults to `int' in declaration of `LOCAL_USER_DECL' app_cepstral.c:44: warning: data definition has no type or storage class app_cepstral.c: In function `cepstral_exec': app_cepstral.c:225: warning: implicit declaration of function `LOCAL_USER_ADD' app_cepstral.c:233: warning: implicit declaration of function `LOCAL_USER_REMOVE' app_cepstral.c: At top level: app_cepstral.c:252: warning: no previous prototype for 'unload_module' app_cepstral.c: In function `unload_module': app_cepstral.c:253: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_cepstral.c: At top level: app_cepstral.c:258: warning: no previous prototype for 'load_module' app_cepstral.c:263: warning: no previous prototype for 'description' app_cepstral.c:268: warning: no previous prototype for 'usecount' app_cepstral.c: In function `usecount': app_cepstral.c:270: warning: implicit declaration of function `STANDARD_USECOUNT' app_cepstral.c: At top level: app_cepstral.c:305: warning: function declaration isn't a prototype make[1]: *** [app_cepstral.o] Error 1 make: *** [apps] Error 2 Eric Hall Vice-president Amaxx, Inc. Customized IT Solutions 5925B Wilcox Place Dublin OH 43016 614.923.6652 - Direct 614.486.3481 - Office 614.923.6652 - eFax Try our off site backup service free for 30 days. blocked::http://www.nationalbackup.com/ http://www.nationalbackup.com blocked::http://www.nationalbackup.com/ ___ The information contained in this message and any attachment may be proprietary, confidential, and privileged or subject to the work product doctrine and thus protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Avaya S8700
Tomer Horn wrote: Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments regarding the matter. The purpose of Asterisk on this matter is to provide outgoing calls from the Avaya through Asterisk, so features such as MWI and stuff are not necessary for me. Thanks, Tomer. I have done it with a Definity G3. It was actually pretty straight forward. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (http://www.neoxo.com) +1 514 395 1106 ext 117 EMail/ MSN ID: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Loosing IAX connection between offices
Setup: Office A: router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv Asterisk: v.1.2.4 static IP Office B: router: Linksys WRT54GL running Linksys firmware v4.30.2 Asterisk: v.1.2.7.1 dynamic IP (using dyndns name) Office A is set up with refresh dns and cron job for iax2 reload every 5 minutes. It rarely looses connection to Office B. Surprisingly, Office B is the one loosing connection with Office A. I'm surprised because Office A is the one with the static IP address. When I do a IAX2 Show Peers, the connection will show as UNKNOWN or UNAVAILABLE. After loosing connection, the only way I can get it to reestablish is to reboot the * box. IAX2 reload doesn't solve it. I haven't been able to establish if it loosing the connection at a specific duration. Though, it seems to be random. iax.conf of Office B: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes notransfer=yes ;(- just added yesterday) [officeb-user] type=user secret=secret host=static ip address context=from-internal [officea] username=officea-user type=peer secret=secret qualify=4000 host=static ip address context=from-internal Any ideas on why Office B is loosing connection to Office A? or how to re-establish connection without rebooting? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura phone does not ring
Fran when you say specify the next hop do you mean the S0 line be an extension in sip.conf or a context in extensions.conf? Or should the line simply be tacked on to my [default] context? Larry Fran Oliveira wrote: I think it is wrong. You should specify the next hop with some like this S0:[EMAIL PROTECTED] 2006/11/23, Larry Alkoff [EMAIL PROTECTED]: Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 registered
On 11/20/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: On 11/20/06, Alex Robar [EMAIL PROTECTED] wrote: Hi Ralph, Have you setup your PAP2 to allow the 729 codec? I believe you actually have to tell it that it's allowed to use that codec before it will work. Cheers, Alex On 11/20/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: Hi guys, I've registered some g729 licenses, during register process everything worked fine. But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk using g729 but I got this answer from asterisk: Should asterisk translate to another codec when trying to make a new call with iax? Why can't asterisk make a call using g729 and sip? Some configuration. Thanks. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn -- Alex Robar [EMAIL PROTECTED] Hi Alex, I set on Audio configuration to enable g729a, g729a as preferred codec and use only preferred codec. Is only that right? With ulaw all calls work fine. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn Hi folks, just to finish this thread. I was trying to call from a pap2 to a pap2, nobody said me tha linksys pap2 can make only one call per time using g729. I tried recording the call to asterisk, another servers and another pap2 and that works fine. Thank you. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)
I am attempting my first go at a simple AGI application using PHP (Getting Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means a professional standard developer. Can't really say what is wrong with your code since I never did an AGI in PHP without this class : http://phpagi.sourceforge.net/ This should make it more easy hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's up with the Manager Interface?!?!
The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Port 5060
That's not possible. These are residential people who hardly know enough to hook up their PAP2 with detailed step-by-step instructions on hand and support on the phone :) Thanks, Daniel -Original Message- From: Tom Lynn [EMAIL PROTECTED] Sent: Wed, November 29, 2006 12:28 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP Port 5060 Can you tunnel through a VPN connection? On 11/28/06, Patrick [EMAIL PROTECTED] wrote: On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote: We have many clients who live in third world countries where the ISPs purposely block traffic on port 5060. I know we could always change the listening port in our Asterisk box. However, doing so will affect all our other users who use port 5060 with no problems. Is there any other solution? I guess I could always run a second instance of Asterisk listening on another port, but is that the cleanest and most scalable solution? Have you tried redirecting the other port with iptables to port 5060 on the Asterisk box? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Port 5060
Wow. I didn't know you could do that. So, I could have something like this in sip.conf: bindport=5060,5080,5081,5082 and it will make Asterisk listen on all those 4 ports? - Daniel -Original Message- From: Joseph [EMAIL PROTECTED] Sent: Wed, November 29, 2006 1:31 am To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP Port 5060 Why not specify, several different ports, in your sip.conf? You can have Asterisk listen on port 5060, 5061, 7080 etc as many as you want; just make sure the port is not taken by some other application. I have 8-phone lines (on sip.conf) and asterisk is listening each line on a different port. -- #Joseph On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote: We have many clients who live in third world countries where the ISPs purposely block traffic on port 5060. I know we could always change the listening port in our Asterisk box. However, doing so will affect all our other users who use port 5060 with no problems. Is there any other solution? I guess I could always run a second instance of Asterisk listening on another port, but is that the cleanest and most scalable solution? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Port 5060
Ok. So, it's not that I can make Asterisk listen on multiple ports for any SIP friend, but I could override the port on an individual SIP friend. So, instead of having something like: bindport=5060,5080,5081,5082 in the general section of sip.conf, I need to just have bindport=5060 in the genearl section and then something like: port=5080 (or whatever number) in any individual SIP friend profile. Correct? Thanks, Daniel -Original Message- From: Zeeshan Zakaria [EMAIL PROTECTED] Sent: Wed, November 29, 2006 4:42 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP Port 5060 Its easy. In sip.conf, under each phone/DID setting, define a different port. e.g. If 8789092323 is assigned to a person in country abc where SIP is blocked, in your sip.conf, under [8789092323] add port=12000 or whatever you like. This will override the default port=5060 setting in sip.conf. I've tried this and it worked for me. But now on the other end the users will also need to change their ports. You'll have to tell them how to change it in their soft/hard phones. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Component::Client::Asterisk::Manager. These have abstracted away the lowest level of programming. I know you've done Python in the past - I hear that there's a module for AMI called py-Asterisk. Have you seen or tried that? Ditto with Ruby - a module called RAMI. Both are on sourceforge. Also, could you hum a few bars about what you're trying to accomplish with your API? I'm curious about the big picture. Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Recording and Call Transfers
Howdy, Anybody have any ideas on how to record to a different file each time a call is transferred by means of the transfer button on Polycom phones? I basically need to be able to execute StopMonitor and an AGI script each time a call is transferred without using features.conf for transfers. Thanks. Stephen Kratzer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3
Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks Eric, I had similar compilation issues when trying to use app_cepstral. This doesn't answer your question, but I've had good success using app_swift. http://www.loopfree.net/app_swift/ Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I am unable to find any included rpms with hudlite...
I am installling on a scratch asterisk running white box linux (fedora) Does anyone know where to find them after the rpm runs. I am looking for ircd and the perl dependancies. The instructions make a ton of assumptions, so I am not sure what is happening here. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's up with the Manager Interface?!?!
Douglas Garstang wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug. ___ just wait till you get a 'hiccup' that causes a line to get cut off, drop a char, and continue on next line. G (examples below) this is an older manager.c there have been a lot of mods to the manager interface in the 1.4 tree, but there is no way i would put that into a production envir. - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 3 Uniqueid: null CallerID: xx CallerIDName: ~308C D13-47426-true~ - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 5 (rest was gone) - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 0 Uniqueid: null CallerID: xx Ca lerIDName: ~308CLD14-40566-true~ - Event: OriginateSuccess Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reaso : 4 Uniqueid: 1163128185.2006 CallerID: xx CallerIDName: ~308CLD13-50454-true~ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting app_cepstral to work with Asterisk 1.4.0-beta3
Eric, It looks like the definition for PTHREAD_MUTEX_RECURSIVE is within an #ifdef __USE_UNIX98 (on Fedora Core 6, anyway). You could try defining it within the Makefile. Similar to the _GNU_SOURCE definition in the app_cepstral.so: app_cepstral.c stanza. Bob... On Wed, 2006-11-29 at 13:38 -0500, Earle Clubb wrote: Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks Eric, I had similar compilation issues when trying to use app_cepstral. This doesn't answer your question, but I've had good success using app_swift. http://www.loopfree.net/app_swift/ Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Component::Client::Asterisk::Manager. These have abstracted away the lowest level of programming. I know you've done Python in the past - I hear that there's a module for AMI called py-Asterisk. Have you seen or tried that? Ditto with Ruby - a module called RAMI. Both are on sourceforge. Also, could you hum a few bars about what you're trying to accomplish with your API? I'm curious about the big picture. Michael, I'm using python. Here's a good example. I'm trying to get SIP blf. I managed to split my result into a list of lines by splitting on ANY of \r\n, \n or \r. I was going use the column headings from the third line as my keys for my dictionary/hash, rather than hard coding them. Notice anything? The 'Call ID' column has a space right in the middle which means I can't simply split this up by white-space. Response: Follows Privilege: Command Peer UserCall ID ExtensionLast state Type xxx.187.128.105 2944090 f7ee98da-6d 2944006 InUse xpidf+xml xxx.187.128.105 2944090 111e388b-6b 2944077 Idle xpidf+xml I think I looked at the python module and was underwhelmed by it. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk connection to a PBX
We are thinking of setting up an Asterisk system to route calls between 2 of our factories. Our idea is to connect an Asterisk box to each PBX and then use SIP(or IAX) to truck between the 2 systems on our internal network. I would be interested in any ideas regarding the connection points: 1. Is using Asterisk a good solution? 2. Is using a T-1 card the best way to connect the PBX and Asterisk? 3. If analog is used for the connection is it better for Asterisk to use FXO or FXS cards? Any ideas are appreciated. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NEED ASTERISK DEVELPER : OH323-asterisk driver and openh323
Dear List, I'm looking for a coder/developer that can modify oh323 return codes on asterisk Example on based on SIP and h323. Right now we are receiving : Call Rejected (code 21) Network Out of Order (code 38) Need to able to replace dose codes with - No Circuit/Channel Available (code 34) Please contact me on [EMAIL PROTECTED] Thanks, Oliver Vermeulen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's up with the Manager Interface?!?!
On 11/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. You would need a ton of rope and a few hundred horses for that :) The Manager API code is distributed across dozens of source files in the Asterisk code base and if you include every possible input and output of it you are talking about code contributions from over a hundred people(including me). The problem is that the Manager API is not centrally coded and has evolved over the last 5 years to accommodate a lot of functionality that it was never designed to do. In my experience, the Manager API is more stable in 1.2 than it was in 1.0, and it was more stable in 1.0 than it was in the pre 1.0 CVS codebase. Having developed a Manager Queue system 3 years ago, I fought a lot of the Manage API battles and eventually got to a fairly robust and non-blocking system that deals a lot of the flaws of the Manager API, but it was by no means a full-featured solution that worked with every kind of Manager input or output. My recommendation to people starting out with Manager API programming is to either specialize it as much as you can to the tasks you need to do with it, or use something like the AstManProxy to handle the connection and interface mess. MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original Message- From: Matt Florell [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What's up with the Manager Interface?!?! On 11/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. You would need a ton of rope and a few hundred horses for that :) The Manager API code is distributed across dozens of source files in the Asterisk code base and if you include every possible input and output of it you are talking about code contributions from over a hundred people(including me). The problem is that the Manager API is not centrally coded and has evolved over the last 5 years to accommodate a lot of functionality that it was never designed to do. In my experience, the Manager API is more stable in 1.2 than it was in 1.0, and it was more stable in 1.0 than it was in the pre 1.0 CVS codebase. Having developed a Manager Queue system 3 years ago, I fought a lot of the Manage API battles and eventually got to a fairly robust and non-blocking system that deals a lot of the flaws of the Manager API, but it was by no means a full-featured solution that worked with every kind of Manager input or output. Agreed on all that, except I found AstManProxy to have just as many issues in the way it rendered XML output. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, SQL ODBC
Hi Norbert - I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a separate SQL box and none of it should affect *. Yes, you can do this. There are other ways to do this without using database storage for voicemail (e.g. you could use normal file-based storage for voicemail and have the webserver connect to the vm storage on the * server via nfs). In my case, that's not possible. Because the box asterisk runs on has very limited memory an no harddisk. I don't know for Peder, but for me the only way seems to be to install ODBC. Just a thought: You could go the other way - share a volume on a separate webserver, and have the asterisk box connect to the webserver via NFS as a client, and store the voicemail on the NFS share. While I don't have any exact numbers, it seems like that would be less overhead (and hassle) than using ODBC/mysql. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
Here's a good example. I'm trying to get SIP blf. I managed to split
my
result into a list of lines by splitting on ANY of \r\n, \n or \r. I
was
going use the column headings from the third line as my keys for my
dictionary/hash, rather than hard coding them. Notice anything? The
'Call
ID' column has a space right in the middle which means I can't simply
split this up by white-space.
Response: Follows
Privilege: Command
Peer UserCall ID Extension
Last
state Type
xxx.187.128.105 2944090 f7ee98da-6d 2944006
InUse
xpidf+xml
xxx.187.128.105 2944090 111e388b-6b 2944077
Idle
xpidf+xml
Cool idea! As far as splitting on whitespace that does look like a
problem. I hate to make assumptions about formatting on output like
this, but perhaps this rule would work: split on at least two
consecutive whitespace chars, assuming that the whitespace between
Peer and User is actually spaces on not tabs. I don't know Python
but the Perl command would look like this:
@keys = split /\s{2,}/, $dataline;
Where @keys is an array of the split out items ('Peer','User','Caller
ID',etc.)
And $dataline is the variable containing the line to be split.
The regexp in the split function says, Split on at least two
consecutive whitespace characters, treating all consecutive whitespace
characters as one delimiter. (Did that make sense?)
I noticed that the output *SEEMS* to be fixed-width, but I think you are
correct about not hard-coding the positions of the fields. If you feel
safe making the assumption that no two fields will ever be separated by
less than two whitespace chars then splitting on the 2-or-more
whitespace chars would work.
BTW, which AMI command returns those results? Just curious.
-MC
P.S. - It's been a learning experience, but the Perl/POE/Asterisk client
combo has been pretty stable. It is event-driven and non-blocking.
Email me off list and I'll be happy to show you what limited success
I've had with it thus far.
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Re: [asterisk-users] No sound: X-Lite - Asterisk - VoIP Provider - Cellphone
Hi Vincent - Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address STUN server : Discover server Port used on local computer : Manually specify range 8000-8019 Here are the ports that I forwarded from my NAT router at home: UDP 5060 UDP 3478 (STUN; needed?) UDP 8000 to 8019 Is there something else I should do, either on my home setup or at work on the NAT router or Asterisk? Just to double check - have you limited the RTP ports on the asterisk server to 8000-8019 (in rtp.conf)? Also, Xlite uses (or used to use) a silence suppresion mechanism that doesn't work too well with asterisk. According to the WIKI: Turn off Silence Supression (to avoid RFC3389 warnings on Asterisk console): Menu | Advanced System Settings | Audio Settings | Silence Settings | Transmit Silence: Yes - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
On Wed, 2006-11-29 at 16:38 +0100, Timothy Parez wrote: I get the following with debug on: P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none P[ 3] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] -- Bearer: Speech P[ 3] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:50556010 oad:497978546 P[ 3] -- CTON: Unknown P[ 3] EXPORT_PID: pid:10 P[ 3] -- PRES: Restricted (0) P[ 3] -- SCREEN: Unscreened (0) Nov 29 16:39:40 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log: Extension can never match, so disconnecting P[ 3] I SEND:RELEASE oad:0497978546 dad:050556010 pid:10 P[ 3] -- bc_state:BCHAN_CLEANED P[ 3] -- channel:1 mode:TE cause:16 ocause:1 rad: cad: P[ 3] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] I IND :RELEASE_COMPLETE oad: dad: pid:10 state:EXTCANTMATCH P[ 3] -- channel:0 mode:TE cause:16 ocause:16 rad: cad: P[ 3] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 3] hangup_chan P[ 3] - hangup P[ 3] * IND : HANGUPpid:10 ctx:inisdn dad:050556010 oad:0497978546 Afaik the dad:050556010 is the destination number and oad:0497978546 is the origination number. I think you need to change your dialplan by adding a 0 in front of your 5055 entries. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, SQL ODBC
Hi Noah, Noah Miller wrote: Hi Norbert - Just a thought: You could go the other way - share a volume on a separate webserver, and have the asterisk box connect to the webserver via NFS as a client, and store the voicemail on the NFS share. While I don't have any exact numbers, it seems like that would be less overhead (and hassle) than using ODBC/mysql. Thanks for your thought. I had this idea too. But having the audio data in an SQL DB has some advantages. And since my setup is a very small one there will be only, lets say, 5 recordings per day on average. (Or maybe even less). So any overhead is no point of discussion. And as I wrote before, Asterisk - mySQl connection is already up and runnig (for CDR). So it just would have been quick and easy if Asterisk could have used the same path for audio data. O.K., lets invest some time in installing ODBC. NOrbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [asterisk-users] Getting app_cepstral to work withAsterisk 1.4.0-beta3
I get an error when I do a make install [EMAIL PROTECTED] app_swift-0.9.5]# make install gcc -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -DCHANNEL_HAS_CID -DNEW_CONFIG -I/opt/swift/include -c -o app_swift.o app_swift.c app_swift.c:49: warning: type defaults to `int' in declaration of `STANDARD_LOCAL_USER' app_swift.c:49: warning: data definition has no type or storage class app_swift.c:50: warning: type defaults to `int' in declaration of `LOCAL_USER_DECL' app_swift.c:50: warning: data definition has no type or storage class app_swift.c: In function `swift_exec': app_swift.c:158: warning: implicit declaration of function `LOCAL_USER_ADD' app_swift.c:162: warning: assignment discards qualifiers from pointer target type app_swift.c:275: warning: implicit declaration of function `LOCAL_USER_REMOVE' app_swift.c: In function `unload_module': app_swift.c:305: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_swift.c:305: error: (Each undeclared identifier is reported only once app_swift.c:305: error: for each function it appears in.) app_swift.c: In function `usecount': app_swift.c:327: warning: implicit declaration of function `STANDARD_USECOUNT' make: *** [app_swift.o] Error 1 [EMAIL PROTECTED] app_swift-0.9.5]# Eric Hall Vice-president Amaxx, Inc. Customized IT Solutions 5925B Wilcox Place Dublin OH 43016 614.923.6652 - Direct 614.486.3481 - Office 614.923.6652 - eFax Try our off site backup service free for 30 days. blocked::http://www.nationalbackup.com/ http://www.nationalbackup.com blocked::http://www.nationalbackup.com/ ___ The information contained in this message and any attachment may be proprietary, confidential, and privileged or subject to the work product doctrine and thus protected from disclosure. If the reader of this message is not the intended recipient, or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Wednesday, November 29, 2006 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] Getting app_cepstral to work withAsterisk 1.4.0-beta3 Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks Eric, I had similar compilation issues when trying to use app_cepstral. This doesn't answer your question, but I've had good success using app_swift. http://www.loopfree.net/app_swift/ Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing streaming MOH in Asterisk
I thought I sent this out.. but don't see it so apologies if it went already. I am trying to get streaming MOH working but haven't been able to.. I am running 1.2.x Based on people's suggestions in other e-mails I've tried: [scanner] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://wgrc.swift-networks.com:8000/ [scanner2] mode=mp3 directory=http://wgrc.swift-networks.com:8000 As well as putting a bogus directory with an MP3 in it in the first one.. nothing. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 601 Second Incoming Call
Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times dosent realize that a new call is coming in. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original Message- From: Douglas Garstang Sent: Wednesday, November 29, 2006 12:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Component::Client::Asterisk::Manager. These have abstracted away the lowest level of programming. I know you've done Python in the past - I hear that there's a module for AMI called py-Asterisk. Have you seen or tried that? Ditto with Ruby - a module called RAMI. Both are on sourceforge. Also, could you hum a few bars about what you're trying to accomplish with your API? I'm curious about the big picture. Michael, I'm using python. Here's a good example. I'm trying to get SIP blf. I managed to split my result into a list of lines by splitting on ANY of \r\n, \n or \r. I was going use the column headings from the third line as my keys for my dictionary/hash, rather than hard coding them. Notice anything? The 'Call ID' column has a space right in the middle which means I can't simply split this up by white-space. Response: Follows Privilege: Command Peer UserCall ID Extension Last state Type xxx.187.128.105 2944090 f7ee98da-6d 2944006 InUse xpidf+xml xxx.187.128.105 2944090 111e388b-6b 2944077 Idle xpidf+xml I think I looked at the python module and was underwhelmed by it. G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : Set MD5Secret: Not set Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HELL is there an asterisk before 'Name'? Now I have to strip the bloody thing out! And why is there TWO empty lines before it? Good grief! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Spandsp rxfax txtax fails no errors
Thanks for the response!!!
I enabled debuging in the menuselect configuration for compiling
asterisk 1.4 beta3. In logging.conf enabled debug loggin to the
/var/log/asterisk/debug file and to the console. Restarted (not just
reload) asterisk and there is plenty of general debugging info in the
debug log file. I also am calling the fax apps with debug argument as
follows
exten = fax,n,rxfax(${FAXFILE}|debug)
and
exten = fax,n,rxfax(${FAXFILE}|debug)
Looking at the code in app_rxfax.c and app_txfax.c there should be
plenty of information in the debug log on failure or success. However I
haven't found any debug log information that should be generated. It is
like it just does a return 0 at the beginning of the application.
I found some documentation on the system() call that says that the dial
plan will jump to n+101 priority if the return value is not 0.
So I setup the dial plan:
[outgoingfax]
exten = out_fax,1,Wait(2)
exten = out_fax,2,txfax(${TXFAX_NAME}|caller|debug)
exten = out_fax,3,system(echo sent fax file ${TXFAX_NAME} /tmp/fax.log )
exten = out_fax,4,Hangup
exten = out_fax,103,system(echo failed fax file ${TXFAX_NAME}
/tmp/fax.log )
exten = h,1,Hangup()
No /tmp/fax.log file created at all.
asterisk -rdddv
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/inettrunk-081e8100, 2)
in new stack
-- Executing [EMAIL PROTECTED]:2] TxFAX(SIP/inettrunk-081e8100,
/tmp/test.tif) in new stack
[Nov 29 13:26:13] DEBUG[28613]: pbx_spool.c:391 scan_service: Delaying
retry since we're currently running '[EMAIL PROTECTED]@ol/asterisk/outgoing/fax.call'
[Nov 29 13:26:24] DEBUG[28613]: pbx_spool.c:391 scan_service: Delaying
retry since we're currently running '/var/spool/asterisk/outgoing/fax.call'
[Nov 29 13:26:35] DEBUG[28613]: pbx_spool.c:391 scan_service: Delaying
retry since we're currently running 'h�,[EMAIL PROTECTED]/asterisk/outgoing/fax.call'
From this it looks like it just gets stuck in the TxFAX app.
Thanks!
On Tue, Nov 21, 2006 at 03:33:57PM -0800, daveasterisk wrote:
Is there anyone who can help with this?
rxfax and txfax when called in the extensions do nothing and no error
are generated that I can find.
I asked something similar on the list a while ago, got no answers and
took a look at the code myself and learned a little bit. Before
I used System(tiff2pdf) to detect errors, which
wasn't so elegant, but worked well anyway.
The description looks like this:
*CLI show application RxFAX
-= Info about application 'RxFAX' =-
[Synopsis]
Receive a FAX to a file
[Description]
RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the
given filename. If the file exists it will be overwritten. The file
should be in TIFF/F format.
The caller option makes the application behave as a calling machine,
rather than the answering machine. The default behaviour is to behave as
an answering machine.
Uses LOCALSTATIONID to identify itself to the remote end.
LOCALHEADERINFO to generate a header line on each page.
Sets REMOTESTATIONID to the sender CSID.
FAXPAGES to the number of pages received.
FAXBITRATE to the transmition rate.
FAXRESOLUTION to the resolution.
Returns -1 when the user hangs up.
Returns 0 otherwise.
If you read the code, a return value of -1 means error and 0
means success, although not clearly stated so in the message
above. So far, that is what you would expect, but return values
are not testable in * dial plans, as far as I know.
I modified app_rxfax.c to set FAXSTATUS to ERROR or SUCCESS and
got it working, but then I discovered that the four return variables
listed above are set only on success. I think that FAXPAGES
would be the best to use for error checking. But still, you
will not get a reason for the failure...
There is a line in the code:
ast_log(LOG_DEBUG, Fax receive not successful - result (%d) %s.\n,
result, t30_completion_code_to_str(result));
that shows us that written information on the type of error *is* available.
These message are in the spandsp code I suppose.
Regards: Håkan
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[asterisk-users] voicemail.conf locking problem
I'm wondering if anyone is having problems when multiple users concurrently change their voicemail passwords. Consider the following scenario (based on vm_change_password() in app_voicemail.c): - user1 wishes to change his password so voicemail.conf is opened and read into a buffer - user1 changes his password - user2 wishes to change his password so voicemail.conf is opened and read into a buffer - voicemail.conf is written with user1's modified password - voicemail.conf is rewritten with user2's modified password but not including user1's modified password because the voicemail.conf that was read by Asterisk when user2 wanted to change his password was read before the changed password of user1 got written back. It seems by looking at the code that this is how it is currently done. The file is not locked down once it is opened. So my question, is the above scenario correct or is there somewhere a lock which I missed out on? Jez Want to start your own business? Learn how on Yahoo! Small Business. http://smallbusiness.yahoo.com/r-index ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
On Wed, 29 Nov 2006, Douglas Garstang wrote: G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : Set MD5Secret: Not set Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HELL is there an asterisk before 'Name'? Now I have to strip the bloody thing out! And why is there TWO empty lines before it? Good grief! Doug. Would it be a better use of your time to fix the offending modules rather than kludge your code to handle the inconsistencies? Is AMI spec'd or would that be the first step? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's up with the Manager Interface?!?!
Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run sip show peer 2944093 in the CLI, you'll see a blank line, followed by a line that is * Name. It appears what you really want is a manager Action to show a sip peer, in which case I would recommend adding a new manager command that returns a string which is much more machine readable. Remember, CLI output is designed to be human readable. Just my $0.02. On 11/29/06 3:36 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Douglas Garstang Sent: Wednesday, November 29, 2006 12:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Component::Client::Asterisk::Manager. These have abstracted away the lowest level of programming. I know you've done Python in the past - I hear that there's a module for AMI called py-Asterisk. Have you seen or tried that? Ditto with Ruby - a module called RAMI. Both are on sourceforge. Also, could you hum a few bars about what you're trying to accomplish with your API? I'm curious about the big picture. Michael, I'm using python. Here's a good example. I'm trying to get SIP blf. I managed to split my result into a list of lines by splitting on ANY of \r\n, \n or \r. I was going use the column headings from the third line as my keys for my dictionary/hash, rather than hard coding them. Notice anything? The 'Call ID' column has a space right in the middle which means I can't simply split this up by white-space. Response: Follows Privilege: Command Peer UserCall ID Extension Last state Type xxx.187.128.105 2944090 f7ee98da-6d 2944006 InUse xpidf+xml xxx.187.128.105 2944090 111e388b-6b 2944077 Idle xpidf+xml I think I looked at the python module and was underwhelmed by it. G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : Set MD5Secret: Not set Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HELL is there an asterisk before 'Name'? Now I have to strip the bloody thing out! And why is there TWO empty lines before it? Good grief! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voicemail.conf locking problem
If you have enough users where this comes up as a real issue, I'd recommend migrating to Asterisk Realtime voicemail, then can have row-level locking etc. if you use the right kind of storage engine... I've found problems using the dial-by-name directory with realtime voicemail, but it seems you might have the scale where some customization work can be justified. Regards, Scott From: [EMAIL PROTECTED] on behalf of je . Sent: Wed 11/29/2006 4:42 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voicemail.conf locking problem I'm wondering if anyone is having problems when multiple users concurrently change their voicemail passwords. Consider the following scenario (based on vm_change_password() in app_voicemail.c): - user1 wishes to change his password so voicemail.conf is opened and read into a buffer - user1 changes his password - user2 wishes to change his password so voicemail.conf is opened and read into a buffer - voicemail.conf is written with user1's modified password - voicemail.conf is rewritten with user2's modified password but not including user1's modified password because the voicemail.conf that was read by Asterisk when user2 wanted to change his password was read before the changed password of user1 got written back. It seems by looking at the code that this is how it is currently done. The file is not locked down once it is opened. So my question, is the above scenario correct or is there somewhere a lock which I missed out on? Jez Want to start your own business? Learn how on Yahoo! Small Business. http://smallbusiness.yahoo.com/r-index ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing software with reseller accounts
I have been using Enswitch. Has some bugs but over all works great. It's not open source but worth the money. - Original Message - From: Guillermo Salas M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 29, 2006 3:12 AM Subject: [asterisk-users] Billing software with reseller accounts Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] beeping noise in background
I have asterisk 1.2.12.1 running with several client phone options. Our echo cancellation is finally working great. The only problem I seem to be having is there is background noise including beeping sounds at regular intervals no matter which phone we use. Does anyone know why? We are using a diqium tdm card. Thanks Kim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original Message- From: Steve Edwards [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! On Wed, 29 Nov 2006, Douglas Garstang wrote: G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : Set MD5Secret: Not set Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HELL is there an asterisk before 'Name'? Now I have to strip the bloody thing out! And why is there TWO empty lines before it? Good grief! Doug. Would it be a better use of your time to fix the offending modules rather than kludge your code to handle the inconsistencies? Is AMI spec'd or would that be the first step? Steve, No... I'm not a C programmer. A standard interface would be a first step. :) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g726 voice prompts
Anyone know if it posible to make voice promps native g726 or g711 format? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What's up with the Manager Interface?!?!
In article [EMAIL PROTECTED], Richard Lyman [EMAIL PROTECTED] wrote: just wait till you get a 'hiccup' that causes a line to get cut off, drop a char, and continue on next line. G (examples below) I've made heavy use of the Manager interface for over 2 years now, and have never seen the kind of behaviour you described and showed examples of. I would be more inclined to suspect the functions you are using to read and collect the AMI output. Perhaps there's a buffer boundary error or something. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modprobe Zaptel
Hi all For some dumb reason I decided to upgrade from Mandriva 2006 to 2007, thinking I could install asterisk all over again. Anyway I did install asterisk, zaptel and libpri. After install I ran modprobe zaptel which said zaptel not found. Thanks to help on this mailing list I had a fix to this problem and edited the Makefile located in /usr/src/linux/ to read -6mdv (instead of -6mdvcustom) which matched the uname -r. Then I installed zaptel again and the drivers were still installed in /lib/modules/2.6.17-6mdvcustom and not /lib/modules/2.6.17-6mdv. After many reinstalls and reboots I could not find why they were still moved to that location, so I just moved them from -6mdvcustom to -6mdv and modprobe zaptel did not display any errors. However I needed to run modprobe wcte11xp for it to actually load the driver, when I did not need to do this in 2006 Does anyone know why this is? I can get it all to start up at boot using rc.local, but when I installed zaptel on Mandriva 2006 it loaded at boot on its own. Still in noob territory, Thanks, Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What's up with the Manager Interface?!?!
-Original Message- From: James Texter [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What's up with the Manager Interface?!?! Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run sip show peer 2944093 in the CLI, you'll see a blank line, followed by a line that is * Name. It appears what you really want is a manager Action to show a sip peer, in which case I would recommend adding a new manager command that returns a string which is much more machine readable. Remember, CLI output is designed to be human readable. James. Ok... that sounds like an objective distinction. Maybe it's just the output that I get as a result of: Action: Command Command: foo that's causing problems. eg: Action: Command Command: sip show subscriptions I don't know why every CLI command doesn't have a corresponding action. I won't be adding any new manager commands, as I am not a C programmer. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manage Users in LDAP
phpldapadmin is pretty nice. I was using 2-3 different ldap clients to get the job done until I got over my php bias and installed it. It lets me do everything I want, without crashing. On 11/27/06, Steven Baker [EMAIL PROTECTED] wrote: Hello All, we are using asterisk+openldap. Do is there any easy way to manage users besides command line or the java ldap browser? -- Check out the all-new Yahoo! Mail betahttp://us.rd.yahoo.com/evt=43257/*http://advision.webevents.yahoo.com/mailbeta- Fire up a more powerful email and get things done faster. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7940 Firmware 8.2
Greetings, I am cutting my teeth with SIP phones and my first issue is getting a Cisco 7940 to Authenticate with my VoIP provider (BBTelsys). I did read some notes on the vo-ip website about 7.5 being the better firmware version. Has anyone had trouble with 8.2 and SIP registering? Should I just downgrade to 7.5 and give it a go? I think SIP uses UDP 5060 correct? The phone is behind a firewall(NAT) I figure this might be an issue as well. Thoughts? Thank you for your response. -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail.conf locking problem
On 17:05, Wed 29 Nov 06, Scott Keagy wrote: If you have enough users where this comes up as a real issue, I'd recommend migrating to Asterisk Realtime voicemail, then can have row-level locking etc. if you use the right kind of storage engine... I've found problems using the dial-by-name directory with realtime voicemail, but it seems you might have the scale where some customization work can be justified. Or use the externpass option in voicemail.conf and write some script/tool to do the passwordchanging for you -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's up with the Manager Interface?!?!
James Texter wrote: Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run sip show peer 2944093 in the CLI, you'll see a blank line, followed by a line that is * Name. It appears what you really want is a manager Action to show a sip peer, in which case I would recommend adding a new manager command that returns a string which is much more machine readable. Remember, CLI output is designed to be human readable. Just my $0.02. action: sippeers or action: sipshowpeer peer: name maybe you should do the below to refresh your memory action: command command: show manager commands ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
