Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
>planning.customer.c...@west-lindsey.co.ukon Thu, 25 Jun 2009 15:11:04 +0800, >Andrew Furey wrote: >On 25/06/2009, David Quinton wrote: >> May be a total red herring (I'm using an old version of Trixbox) >> but if I edit my PHPs on a Windows machine and upload using FTP, they >> will only run if I fire up Nano and save the file on the Asterisk box. > >I haven't used TrixBox, but that sounds a lot like CRLF<->LF issues to me... I agree. But it doesn't happen with .php's run by Apache. It's just something I live with! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls dropping
Hi,
I am using a call file formated like this:
Channel: local/12125557...@outbound/n
Callerid: 12125551212
Context: detect
Extension: s
Priority: 1
This sends the call into the dialplan at the [outbound] context. In
[outbound], I have:
[outbound]
exten => _1.,1,Dial(SIP/${ext...@flowroute,43)
If the call is answered, it move on to the [detect] context.
When using this method, it appears that the call file creates the first part
of the call, then creates a second call with the Dial() app. Once the call
executed by the Dial() app is answered, the two calls are joined together.
What I am experiencing is that sometime the first part of the call drops and
therefore is never joined to the second part of the call. I see errors like
this:
== Spawn extension (outbound, 12125557891, 1) exited non-zero on
'Local/12125557...@outbound-8392;2'
[Jun 25 15:58:09] ERROR[7272]: pbx.c:8637 device_state_cb: Received invalid
event that had no device IE
[Jun 25 15:58:09] ERROR[7272]: app_queue.c:810 device_state_cb: Received
invalid event that had no device IE
I am running Asterisk 1.6.1.1
If more info is needed, please let me know.
Thanks for the help!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_cepstral, register & existing Cepstral licenses.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin P. Fleming wrote: > There is no need; your existing Cepstral-supplied licenses will continue > to operate, and will be added to any Digium-supplied licenses you > purchase and activate. Thanks Kevin. So I shouldn't worry about this? corp-asterisk*CLI> cepstral show licenses Cepstral Licensing Information == Allison Voice Enabled: no Total licensed ports: 0 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKQ/M8CFu3bIiwtTARAui1AJ48Jf/E0gVGrITS32SHiZSCIXfczgCeMIyi 4M9g7qv/k7Vrxy/mJA4j3kA= =5YbR -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Persistent dynamic queue members
Miguel Molina escribió: Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue members are loaded /before/ the agents.conf file, so the agent dynamic queue members stay invalid until the Queue application is reloaded. - Is this a normal behavior? or am I missing something? Is there a way to workaround it? - Is someone else running into this? - Should I report this in the bugtracker? Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center I found a solution for this, preload => chan_agent.so on modules.conf. Sorry for the noise. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_cepstral, register & existing Cepstral licenses.
Barry L. Kline wrote: > I'm going to end up buying more ports from Digium but I'd like to also > use the existing voice/port licenses that I currently have. Is this > possible? Is there anyway to migrate the licenses to the Digium > implementation of Cepstral? There is no need; your existing Cepstral-supplied licenses will continue to operate, and will be added to any Digium-supplied licenses you purchase and activate. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Persistent dynamic queue members
Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue members are loaded /before/ the agents.conf file, so the agent dynamic queue members stay invalid until the Queue application is reloaded. - Is this a normal behavior? or am I missing something? Is there a way to workaround it? - Is someone else running into this? - Should I report this in the bugtracker? Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registration fails
SIP-registration errors are solved by restarting the Asterisk-server.
But I expect them to return in time...
I can make call now, but the other end does not hear me. So problem with
RTP-flow...
Can someone guide me to the solution ?
On the Asterisk-server I have this (iptables):
-A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT
-A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 25 -j
ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j
ACCEPT
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited
In rtp.conf I have this :
rtpstart=11000
rtpend=11500
Asterisk is behind firewall. Endian firewall has following
configuration :
enable SIP proxy transparant
RTP port low : 11000
RTP port high : 11500
Firewall port forwarding : uplink:5060 >>> asterisk_ip:5060
Asterisk himself says :
-- Executing [050510...@intern:1] NoOp("SIP/grandstream-09813b58",
"via 3StarsNet") in new stack
-- Executing [050510...@intern:2] Dial("SIP/grandstream-09813b58",
"SIP/3starsnet/050510484") in new stack
-- Called 3starsnet/050510484
-- SIP/3starsnet-0981bf08 is making progress passing it to
SIP/grandstream-09813b58
-- SIP/3starsnet-0981bf08 answered SIP/grandstream-09813b58
== Spawn extension (intern, 050510484, 2) exited non-zero on
'SIP/grandstream-09813b58'
What do I need in sip.conf to overcome these rtp-problems ??
I have :
externip=78.21.62.99
canreinvite=no
jbenable = yes
[3starsnet]
type=peer
...
nat=yes
...
Thanks for the help !
Jonas.
On Thu, 2009-06-25 at 17:25 +0200, jonas kellens wrote:
> Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
> opened and 5060 forwarded to Asterisk (192.168.2.2)
>
> Can someone see why SIP-registration fails ??
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_cepstral, register & existing Cepstral licenses.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a license for Allison-8kHz and two concurrent port licenses that I purchased from Cepstral at the end of last year. I just got around to installing to my * 1.6.0.10 machine. I've decided that the best way for me to integrate the two would be res_cepstral, which I downloaded and installed. Everything is fine, except the register program, which is looking for a license key sent from Digium. I'm going to end up buying more ports from Digium but I'd like to also use the existing voice/port licenses that I currently have. Is this possible? Is there anyway to migrate the licenses to the Digium implementation of Cepstral? TIA, Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKQ9RICFu3bIiwtTARAjg4AKCriHJ0F18y7HJvbby9FjbCWL72OQCfW3Cy SipgbgQvZb93O3u4ecsxxCY= =9bF7 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PHP AGI Not Working and Odd Behavior
I think I found the part of my AGI the script is stuck at.
The #!/usr/bin/php command was fine. What the agi debug I believe is
displaying is the output of this:
$in = fopen("php://stdin","r");
Which explains what I thought was "cached"
-- the same #!/usr/bin/php5 -q command repeatedly failing was NOT the
opening of my script, but rather the
I'm going to play around more with this, but in case anyone has a
similar problem, they'll have a starting place.
Thanks!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 59, Issue 62 (Should be my PHP/AGI problem and odd behavior)
Thanks...completely irrelevant in my case though. I installed and set up php on my end, so I have short_open_tag to be on :) Any other ideas anyone? Leah Newmark VoIP Programmer Capalon Communications _ Probably unrelated, but its bad practice to use short tags / use / instead. Incase short_open_tag = Off in php.ini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 59, Issue 62
Probably unrelated, but its bad practice to use short tags / use / instead. Incase short_open_tag = Off in php.ini On Thu, Jun 25, 2009 at 11:08 AM, Leah Newmark wrote: > Take a look at this: > /var/lib/asterisk/agi-bin/olehphone# head incoming.php displays: > #!/usr/bin/php > > Running it shows this: > /var/lib/asterisk/agi-bin/olehphone# ./incoming.php > #!/usr/bin/php5 -q > > > Is that normal behavior if php5 is the library installed? Seems very odd > to me. > > LN > > > asterisk-users-requ...@lists.digium.com wrote: >> Message: 18 >> Date: Wed, 24 Jun 2009 16:23:17 -0500 >> From: "Danny Nicholas" >> Subject: Re: [asterisk-users] PHP AGI Not Working and Odd Behavior >> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" >> >> Message-ID: <6e120f20f06549dabcbf9675d32cb...@db0002> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Looking at my ?man php5? ?q is not a valid option. That may be just on >> Suse. >> >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Assigning an IVR to an extension in *NOW 1.5/FreePBX
Hi all, On my *NOW beta box, I was able to assign an IVR an extension (801, etc.) With the new 1.5 *NOW and freepbx, I can create an IVR, but how do I assign it an extension, so I can dial or transfer users to that IVR? Thanks in advance for the help. I checked the freepbx docs but didn't find any answers. --Zaheer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing line 2 from CISCO 7940g
David's directions will work on a 7941/7961, not the 7940/7960. You do have
to keep the line configuration for the 79x0 series phones in the
SIP${MAC}.cnf file.. I have not tested setting them to "", but I know if
you telnet into the phone they will show "UNPROVISIONED" as the setting.
You can also clear all of the cached settings by telnetting into the phone,
clearing the config, and resetting it.
-Jonathan
On Thu, Jun 25, 2009 at 8:29 AM, David Gibbons wrote:
> Mike,
>
> 1. Remove the 'line 2' entries completely from the SEPXX.XML file.
> 2. Change the 'Version' tag in the SEPXX.XML file. You need only
> change one digit; I usually just increment the last digit.
> (1.0.0.0-9).
> 3. Restart the phone (Settings -> **#**).
> 4. This should do it. If it doesn't, proceed to step 5 with caution.
> 5. If the line still appears, reset the phone to factory defaults
> (Hold # while booting, then dial 123456789*0# when the line lights flash
> amber back and forth). DO NOT RESET TO FACTORY DEFAULTS IF YOU DON'T HAVE
> THE TFTP SERVER SETUP WITH THE FIRMWARE IMAGES. This will force the phone to
> re-download the SEP.XML file.
>
> -Dave
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
> Sent: Wednesday, June 24, 2009 5:12 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Removing line 2 from CISCO 7940g
>
> Folks,
>
> I have CISCO 7940g phone. I have in the past configured the phone with two
> lines. Having found the 2nd line wasn't much use, I want to remove it from
> the config. I have taken it out of the SIP config file that is TFTPd to the
> phone but it is still showing on the phone and it is still trying to log
> into Asterisk with that account. I have tried removing the config line and
> blanking out the options but it still persists.
> Does anyoen know how to get rid of the thing?
>
> Mike.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing line 2 from CISCO 7940g
Mike, 1. Remove the 'line 2' entries completely from the SEPXX.XML file. 2. Change the 'Version' tag in the SEPXX.XML file. You need only change one digit; I usually just increment the last digit. (1.0.0.0-9). 3. Restart the phone (Settings -> **#**). 4. This should do it. If it doesn't, proceed to step 5 with caution. 5. If the line still appears, reset the phone to factory defaults (Hold # while booting, then dial 123456789*0# when the line lights flash amber back and forth). DO NOT RESET TO FACTORY DEFAULTS IF YOU DON'T HAVE THE TFTP SERVER SETUP WITH THE FIRMWARE IMAGES. This will force the phone to re-download the SEP.XML file. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 24, 2009 5:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Removing line 2 from CISCO 7940g Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP registration fails
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:x...@85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret= fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes nat=yes disallow=all allow=gsm allow=alaw [Jun 25 16:54:32] NOTICE[32550]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #54) Really destroying SIP dialog '628e05295c1a2cc560d1c6c073b85...@127.0.0.1' Method: REGISTER Really destroying SIP dialog '4f9b2b7a241f3f2a193ceb0020778...@192.168.2.2' Method: OPTIONS Retransmitting #4 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK15be023b;rport From: ;tag=as36b44350 To: Call-ID: 628e05295c1a2cc560d1c6c073b85...@127.0.0.1 CSeq: 156 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #5 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK15be023b;rport From: ;tag=as36b44350 To: Call-ID: 628e05295c1a2cc560d1c6c073b85...@127.0.0.1 CSeq: 156 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #3 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 59, Issue 62
The script runs fine command line. I have edited in the past to try as /usr/bin/php -q and it didn't help. Right now, it's not even reading the changes. I must be missing something very obvious... LN asterisk-users-requ...@lists.digium.com wrote: > Message: 16 > Date: Wed, 24 Jun 2009 17:17:59 -0400 > From: "Juan E. Rodr?guez" > Subject: Re: [asterisk-users] PHP AGI Not Working and Odd Behavior > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <4a429807.5060...@gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Try running your script with /usr/bin/php5 script.php to test it > Or changing #!/usr/bin/php5 -q to #!/usr/bin/php -q > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 59, Issue 62
Take a look at this: /var/lib/asterisk/agi-bin/olehphone# head incoming.php displays: #!/usr/bin/php Message: 18 > Date: Wed, 24 Jun 2009 16:23:17 -0500 > From: "Danny Nicholas" > Subject: Re: [asterisk-users] PHP AGI Not Working and Odd Behavior > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Message-ID: <6e120f20f06549dabcbf9675d32cb...@db0002> > Content-Type: text/plain; charset="iso-8859-1" > > Looking at my ?man php5? ?q is not a valid option. That may be just on > Suse. > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
Thanks for the suggestion, but I'm editing directly on the server I've been doing AGIs for, what, 4 years now? I have never been *this* stumped! __ On Wed, 24 Jun 2009 15:43:14 -0400, Leah Newmark http://lists.digium.com/mailman/listinfo/asterisk-users>> wrote: /I also have noticed odd behavior. When I edit an AGI, the changes aren't />/always showing up in the running of the script via asterisk. / May be a total red herring (I'm using an old version of Trixbox) but if I edit my PHPs on a Windows machine and upload using FTP, they will only run if I fire up Nano and save the file on the Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI Transfer?
Trying to accomplish something that seems simple enough but I've tried
everything I can think of but I cannot get an AGI Transfer to work.
Seems simple enough
$AGI->exec('transfer','SIP/101');
and here's the resultant Debug:
AGI Rx << EXEC transfer "SIP/101"
-- AGI Script Executing Application: (transfer) Options: (SIP/101)
AGI Tx >> 200 result=0
But nothing happens.
I can't use "dial" (which does work incidentally) because "dial" creates
channels and messes with the CDR info.
Any ideas?
Thanks!
PB
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing line 2 from CISCO 7940g
You have to still have all of the line2 entries in the config file and they have to be set to "". -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 24, 2009 4:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Removing line 2 from CISCO 7940g Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
Why ask for a GUI ? --> I do not ask for one... my customer does. Realtime (did some reading) + custom webinterface seems indeed one of the best solutions. Then the thing is that I am not a programmer. I am a graduated network engineer with affection towards VoIP and OpenSource (Asterisk). I do know PHP + MySQL... Will have to do some studying then... On Thu, 2009-06-25 at 11:40 +, Jeff LaCoursiere wrote: > > On Thu, 25 Jun 2009, jonas kellens wrote: > > > I feel a great preference for sticking to manually editing > > the .conf-files. > > Then why did you ask for a GUI? > > > But if I define in the contract that changes to the > > Asterisk-PBX need to be done by me, I force a maintenance cost towards > > the customer and that is not always what is requested... > > > > You have been given lots of good suggestions. The best, IMO, is for you > to use realtime, and make your own web interface for your customers. > > j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Jabber
jonas kellens wrote: > So, what about this 'iksemel' ?? On RedHat based distros, you can install EPEL and the Dag Wieers repos in order to get extra things like iksemel. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hotdesk and voicemail
Julian Lyndon-Smith wrote:
> We have several types of phones,
>
> cisco 7940/7960
> aastra 55i/9113i/
> grandstream gxp2010
>
>
> I want to be able to implement hotdesking where an agent will logon to
> any phone. I got all of that working, without having to reboot phones,
> but then hit a brick wall.
>
> Voicemail.
>
> I still want each phone to use the BLF for voicemail indication, and to
> use the "voicemail button" to dial voicemail directly. Is it possible to
> do this dynamically, or will I have to rewrite the phone config and reboot ?
>
> The issue I have with rebooting is that the cisco's take so bloody long
> to reboot (mainly waiting around at the VLAN) that it is unusable. Does
> anyone have any solutions to make the VLAN problem go away ? We don't
> use cisco switches.
What you'll have to do is use the database (AstDB, MySQL, or whatever you're
using to track the hotdesking portion), and configure the phones to dial the
Voicemail() application with a general extension (for example, 8500, default on
Cisco).
When you dial that, you'll need to do some pre_voicemail extension that will
determine what device is requesting Voicemail() (i.e. ${CHANNEL(peername)} --
at
least in the 1.6.2 branch, otherwise you'll have to use something like
${CUT(CUT(CHANNEL,/,2),-,1)} ). Then when you have that, you can lookup what
extension number/voicemail box you are trying to dial, based on your logged in
status.
With 1.6.2, you can use the MinivmMWI() to enable/disable the MWI for devices
as
your agents login and logout, and whenever you finish calling Voicemail() or
VoicemailMain().
I have recently built a full system for a client using these methods, and will
be writing an article on it over the next couple of weeks (as soon as I find
some free time to write it!).
Good luck!
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxclient softphone: quality?
Hi All; Did anyone used iaxclient? I would like to know how is the voice quality? OS to be used Microsoft. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
On Thu, 25 Jun 2009, jonas kellens wrote: > I feel a great preference for sticking to manually editing > the .conf-files. Then why did you ask for a GUI? > But if I define in the contract that changes to the > Asterisk-PBX need to be done by me, I force a maintenance cost towards > the customer and that is not always what is requested... > You have been given lots of good suggestions. The best, IMO, is for you to use realtime, and make your own web interface for your customers. j > > On Thu, 2009-06-25 at 15:26 +0530, Jaswinder Singh wrote: > >> If you plan it right from the start, FreePBX can save hell lot of >> time. Instead of fixing in include files, you can also create custom >> contexts from within the GUI now, i am sure there is a module for that >> as well. As said above, either stick fully to GUI or fully to manual >> configurations. Ugly mixing of the two will definitely bite you later >> on if you don't know what you are doing . > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote: > In FreePBX there are whatever_custom.conf files that are not touched when > changes are made in the GUI. A GUI for Asterisk does not necessrily imply FreePBX. There are certainly other ways to do that. For instance: * asterisk-gui: Uses Asterisk configuration files as its DB. Edits them through the the manager interface (manager-over-http). In many cases you can just edit config files and your changes will remain in tact. A pain to debug, and mixes data and code in ugly ways, though. * Druid: http://voiceroute.org/ - mixes editing config files and storing configuration in a database. Much saner data model than FreePBX. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
Asterisk RealTime is perfect for this as you can built them a simple web interface to make insertions into your sip table. Ish jonas kellens wrote: > Tzafrir Cohen, > > if mixing hand-written configs with GUI-configs is not 'good > practise', then how to build a scalable Asterisk IP-PBX where the > customer is not 100% dependent of the implementer ? > > Like I already said, I got the remark "To add a new phone, I do not > want to be forced to call you". And I don't see a CEO of a > meat-company learning some vim-skills... > > I don't know how to put "the simpler administration" into the hands of > a noob, without me having to put a 100% support into the contract > (which is overkill). > > Jonas. > > > On Wed, 2009-06-24 at 23:39 +0300, Tzafrir Cohen wrote: >> On Wed, Jun 24, 2009 at 09:20:44PM +0200, jonas kellens wrote: >> > I wonder if there is a GUI that does not change the underlying hand-made >> > configuration ?! >> > >> > What I'm looking for actually is a GUI for adding a new SIP-client + >> > voicemail, so that a company does not have to call me when they hired a >> > new employee. >> > >> > I don't want a GUI that over-writes my hand-made SIP-configuration, and >> > my hand-made dialplan. >> >> You're looking at it the wrong way. Figure out where the GUI generates / >> updates the configuration and make sure it gets things right. >> >> Either you write configuration manually or the GUI writes them. Don't >> try mixing both too badly. >> >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
I feel a great preference for sticking to manually editing the .conf-files. But if I define in the contract that changes to the Asterisk-PBX need to be done by me, I force a maintenance cost towards the customer and that is not always what is requested... On Thu, 2009-06-25 at 15:26 +0530, Jaswinder Singh wrote: > If you plan it right from the start, FreePBX can save hell lot of > time. Instead of fixing in include files, you can also create custom > contexts from within the GUI now, i am sure there is a module for that > as well. As said above, either stick fully to GUI or fully to manual > configurations. Ugly mixing of the two will definitely bite you later > on if you don't know what you are doing . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
If you plan it right from the start, FreePBX can save hell lot of time. Instead of fixing in include files, you can also create custom contexts from within the GUI now, i am sure there is a module for that as well. As said above, either stick fully to GUI or fully to manual configurations. Ugly mixing of the two will definitely bite you later on if you don't know what you are doing . On Thu, Jun 25, 2009 at 2:41 PM, Tzafrir Cohen wrote: > On Thu, Jun 25, 2009 at 11:02:44AM +0200, jonas kellens wrote: > > Tzafrir Cohen, > > > > if mixing hand-written configs with GUI-configs is not 'good practise', > > then how to build a scalable Asterisk IP-PBX where the customer is not > > 100% dependent of the implementer ? > > > > Like I already said, I got the remark "To add a new phone, I do not want > > to be forced to call you". And I don't see a CEO of a meat-company > > learning some vim-skills... > > > > I don't know how to put "the simpler administration" into the hands of a > > noob, without me having to put a 100% support into the contract (which > > is overkill). > > It means you should adapt the said GUI to generate the right > configuration. > > -- > Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hotdesk and voicemail
We have several types of phones, cisco 7940/7960 aastra 55i/9113i/ grandstream gxp2010 I want to be able to implement hotdesking where an agent will logon to any phone. I got all of that working, without having to reboot phones, but then hit a brick wall. Voicemail. I still want each phone to use the BLF for voicemail indication, and to use the "voicemail button" to dial voicemail directly. Is it possible to do this dynamically, or will I have to rewrite the phone config and reboot ? The issue I have with rebooting is that the cisco's take so bloody long to reboot (mainly waiting around at the VLAN) that it is unusable. Does anyone have any solutions to make the VLAN problem go away ? We don't use cisco switches. TIA Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
On Thu, Jun 25, 2009 at 11:02:44AM +0200, jonas kellens wrote: > Tzafrir Cohen, > > if mixing hand-written configs with GUI-configs is not 'good practise', > then how to build a scalable Asterisk IP-PBX where the customer is not > 100% dependent of the implementer ? > > Like I already said, I got the remark "To add a new phone, I do not want > to be forced to call you". And I don't see a CEO of a meat-company > learning some vim-skills... > > I don't know how to put "the simpler administration" into the hands of a > noob, without me having to put a 100% support into the contract (which > is overkill). It means you should adapt the said GUI to generate the right configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk
Tzafrir Cohen, if mixing hand-written configs with GUI-configs is not 'good practise', then how to build a scalable Asterisk IP-PBX where the customer is not 100% dependent of the implementer ? Like I already said, I got the remark "To add a new phone, I do not want to be forced to call you". And I don't see a CEO of a meat-company learning some vim-skills... I don't know how to put "the simpler administration" into the hands of a noob, without me having to put a 100% support into the contract (which is overkill). Jonas. On Wed, 2009-06-24 at 23:39 +0300, Tzafrir Cohen wrote: > On Wed, Jun 24, 2009 at 09:20:44PM +0200, jonas kellens wrote: > > I wonder if there is a GUI that does not change the underlying hand-made > > configuration ?! > > > > What I'm looking for actually is a GUI for adding a new SIP-client + > > voicemail, so that a company does not have to call me when they hired a > > new employee. > > > > I don't want a GUI that over-writes my hand-made SIP-configuration, and > > my hand-made dialplan. > > You're looking at it the wrong way. Figure out where the GUI generates / > updates the configuration and make sure it gets things right. > > Either you write configuration manually or the GUI writes them. Don't > try mixing both too badly. > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
On 25/06/2009, David Quinton wrote: > May be a total red herring (I'm using an old version of Trixbox) > but if I edit my PHPs on a Windows machine and upload using FTP, they > will only run if I fire up Nano and save the file on the Asterisk box. I haven't used TrixBox, but that sounds a lot like CRLF<->LF issues to me... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
On Wed, 24 Jun 2009 15:43:14 -0400, Leah Newmark wrote: >I also have noticed odd behavior. When I edit an AGI, the changes aren't >always showing up in the running of the script via asterisk. May be a total red herring (I'm using an old version of Trixbox) but if I edit my PHPs on a Windows machine and upload using FTP, they will only run if I fire up Nano and save the file on the Asterisk box. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
