Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910
Javier Chia ha scritto: The phone is now logged in but can´t place nor receive calls. It keeps giving Busy tone when I try to dial a number. it does happen when there are no matching extensions for the number you are dialing internal context is ok? you can dial just internal context extensions from the 7910 But in: Asterisk*CLI shows the following: please from the cli: sccp debug 10 place a call from the cisco 7910 place a call from X-lite to the cisco 7910 post the log so I can see what is wrong Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Receiving fax by app_rxfax over h.323 trunk
Hi,
does anybody has working this konfiguration? For me app_rxfax start receiving,
fax start sending, but after few seconds at begining of the page it stop with
error 400.
My HW PBX configuration is:
ISDN PRI - AVAYA S8300 - H.323 channel - * with app_rxfax
My extensions.conf is:
'7406211' = 1. Goto(fax|666|1)
[fax]
'666' = 1. SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
2. rxfax(${FAXFILE})
'h' = 1. system(/usr/sbin/mailfax ${FAXFILE} [EMAIL PROTECTED] ${CALLERIDNUM})
Thanks for help,
Bob.
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Re: [Asterisk-Users] McLeod Integrated T1 - no PRI?
Does anyone have any ideas? Any magic words to give to the people at McLeod to get this running? You might ask the carrier to take a careful look at the mapping for the d-channel in their DACS equipment and perhaps even ask them to try re-mapping it for you. If that does not get things moving, then I would ask them to send somebody onsite with a PRI test set and see if they can complete any test calls with it. In any event, you should set your configuration to clock from the T1 circuit to avoid timing slips - although I don't expect it is the root cause of this particular problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SP300 config files
Hi, all Sorry for not exactly on-topic question I got Polycom SP300 phones. Somehow they did not come with software. I will call them on Monday, but in the meantime, I would like to get them going. I need Polycom configuration template files (phone.cfg, sip.cfg and whatever else they supply). Did not find them on the Polycom site. Can someone e-mail those to me ([EMAIL PROTECTED])? Then I will be able to work on the weekend. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About the using of astmanproxy
Hi,I met the same problem as this mail,
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg101451.html
***Hello,
I want to recieve the output from astmanproxy in a php script.
Is that possible ?
I made a simple php script:
PRE
?php
$socket = fsockopen("127.0.0.1","1234", $errno, $errstr, $timeout);
fputs($socket, "Action: Login\r\n");
fputs($socket, "UserName: xxx\r\n");
fputs($socket, "Secret: xxx\r\n\r\n");
fputs($socket, "Action: Command\r\n");
fputs($socket, "Command: Show Channels\r\n\r\n");
fputs($socket, "Action: Logoff\r\n\r\n");
while (!feof($socket)) {
$wrets[] = fread($socket, 8192);
}
fclose($socket);
var_dump($wrets);
?
/pre
Output in debugmode at the console is correct, but I cannot read the output in php. If I use port 5038 I get the output, but I want to connect with multiple clients, so I should't use a direct connection to manager api, right ? Why can't I read the output from astmanproxy ?
--
Regards
***
I searched the archive, and didn't find the answer.
Anyone knows how to solve it ?
and Christian,may you not mind me borrowingur mail content.
BTW: Do u get the answer?
Thanks a lot!
Best Regards,
Gary Li__赶快注册雅虎超大容量免费邮箱?http://cn.mail.yahoo.com___
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[Asterisk-Users] how to edit ring time
i dont how to edit the time for ringing 3ms to 4ms when it displayed on console Nobody picked up in 3 ms and its very short time for ringing . please if anyone can help me do it please. Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXphone - ip address - extension number.
Tzafrir Cohen wrote: On Fri, Jul 08, 2005 at 11:12:37PM +0200, Zoltan Szecsei wrote: Is this how the modprobes are supposed to respond?? gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # If you use 2.4, consider 2.6, as its ztdummy works better. If you use 2.6, you may be using udev, and need to read README.udev . Bingo! I had read the README.udev, but had not noticed any make-time udev related messages so chose to ignore its contents. Bad, bad boy - naughty me. Anyway, dumping those lines into the 50-udev-rules file has solved this issue perfectly. I can now modprobe zaptel then ztdummy with no errors and asterisk loads opens all the IAX stuff confirms that it is listening on port 4569. Cool - well spotted. Now to get rid of these darn issues: *CLI Jul 9 10:23:46 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! -- Accepting unauthenticated call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, IAX/z2|20|tr) in new stack Jul 9 10:23:57 WARNING[8102]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 10:23:57 NOTICE[8102]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 10:24:07 WARNING[8102]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/5' Jul 9 10:24:18 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:24:20 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 I'm sure its now possibly a phone setup, but being green at this asterisk stuff, I let them sell me the phones without setup manuals or anything. grrr. In response to the ethereal queries I was asked: All PCs phones on this network plug into the same 3com switch, port 8 of this switch links to a netgear DG834 (my ADSL modem/router/firewall). The three remaining ports on the netgear are unused, but if I plug anything into them, the netgear DHCP offers them a compatible IP and they can see everything plugged into the 3COM. Note that *all* equipment that is permanently connected has a *fixed* IP and does not get one from the netgear DHCP service. In short, the fact that ethereal saw nothing coming from the phones suggest a phone setup issue. However the fact that when I dial *1202 (*1 being the phones IAX dial-prefix), asterisk responds to the request, shows that the phones do however communicate in some recognisable way. Maybe I should also fiddle with a minimum iax.conf and extensions.conf. Chat soon thanks, Zoltan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SP300 config files
http://www.freedomphones.net/polycom/files/ On Sat, 2005-07-09 at 17:03 +1000, Rudolf Ladyzhenskii wrote: Hi, all Sorry for not exactly on-topic question I got Polycom SP300 phones. Somehow they did not come with software. I will call them on Monday, but in the meantime, I would like to get them going. I need Polycom configuration template files (phone.cfg, sip.cfg and whatever else they supply). Did not find them on the Polycom site. Can someone e-mail those to me ([EMAIL PROTECTED])? Then I will be able to work on the weekend. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Closest dialplan language equivalent for dialparties.agi ?
Hi, I'm using AMP and its dialparties.agi as most important script in system. I'd like to port configuration to more embedded system, where I don't have Perl available. So I'd like to implement dialparties.agi functionality as closest as possible with dialplan language. Are there any existing dialplan scripts-examples that are close related to dialparties.agi functionality ? Is it possible to use compiled Perl AGI script in binary form also as AGI script ? How to ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to edit ring time
wassim darwish wrote: i dont how to edit the time for ringing 3ms to 4ms when it displayed on console Nobody picked up in 3 ms and its very short time for ringing . please if anyone can help me do it please. This is now a joke, right? B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Hi, I'm not sure if DTMF is convenient solution for user that has cellular on his ear Regards, Rob. - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Ed Greenberg [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 08, 2005 4:15 PM Subject: RE: [Asterisk-Users] Speech Recognition Ed can I ask you a question, Not trying to influence you one way or the other but why deal with the 'issues' of speech recognition when what you are looking to achieve is easily met with dtmf codes. Dtmf, works, is easy to manage and well established. Speech should only be used when you need to enter complex controls with more than '9' easy options etc. Just a thought. Cheers, Dean -Original Message- From: Ed Greenberg [mailto:[EMAIL PROTECTED] Sent: Friday, 8 July 2005 9:32 AM To: Dean Collins; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Speech Recognition Tell me probably is excessive. I just really need to recognize Yes, No, One, Two, Three and Four. The Sphinx suggestion should help though. /edg --On Friday, July 08, 2005 8:27 AM -0400 Dean Collins [EMAIL PROTECTED] wrote: Hi Ed, Did you read the wiki comment on Tellme? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to edit ring time
On 09/07/05, wassim darwish [EMAIL PROTECTED] wrote: i dont how to edit the time for ringing 3ms to 4ms when it displayed on console Nobody picked up in 3 ms and its very short time for ringing . please if anyone can help me do it please. Didn't any of the 5 answers you got to your last posting solve the problem? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] make available again a zap channel after a red alarm...
Hello, I finally arrived to convince a cellsocket for Nokia phones to work with a X101P card in an asterisk v1.0.7. The problem I have now is that cellsocket usually resets after receiving a call in the mobile. If asterisk by luck notices it, it issues an error message Detected Alarm on channel 4: Red Alarm and makes unavailable the Zap channel. The problem is solved by removing and puting back the cable, but this is not a solution for my case. 1. Is there a way to make asterisk to stop monitoring the status of the line? 2. Is there a way by software to make available again the Zap channel after a red alarm ? Thank you in advance, Dimitris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
- Original Message - From: Richard Koch [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 08, 2005 4:38 PM Subject: [Asterisk-Users] Speech Recognition Ed, Check this out: http://turnkey-solution.com/asterisk-sphinx.html That got me up in running in no time. -Rick What are you experiences with recognition accuracy and user acceptance ? Any more info you're willing to share will help out others Regards, Rob. -Original Message- From: Ed Greenberg [mailto:edg at greenberg.org] Sent: Friday, 8 July 2005 9:32 AM To: Dean Collins; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Speech Recognition Tell me probably is excessive. I just really need to recognize Yes, No, One, Two, Three and Four. The Sphinx suggestion should help though. /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phone w/ XML browser
Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so any other recommendation? Seems, that xml minibrowser isn't obvious even in high end phone, but I think that via this function can be phone very extensible... thanks PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Definitive CallerID Format and anonymous?
Thanks, Mark. I've changed several of my old Set entries, but
totally spaced out on that one. Done now though. :)
The new format is:
exten = _1NX,1,Set(CALLERID(number)=4025551212|a)
exten = _1NX,2,Set(CALLERID(name)=NPI|a)
exten = _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
MARK.
Rich Adamson wrote:
Thanks for the thorough reply.
I'm aware that there necessarily are inconsistencies between termination
providers; I was just curious to find out if there's some form of
standard one should follow, which may either result in more consistent
behavior, or at least shift culpability to the provider...
Bottom line is that I want to tell my IAX providers Hey, I'm doing the
right thing, could you find out where it's breaking?
Here's what I've been told to use and it works with teliax Nufone:
exten = _1NX,1,SetCallerID(4025551212|a)
exten = _1NX,2,SetCIDName(NPI|a)
exten = _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
The setcidname isn't really needed under most circumstances, but I
send it anyway for consistency.
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Re: [Asterisk-Users] Polycom SP300 config files
Thanks, Rudolf - Original Message - From: Scott Kamp [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 10, 2005 1:00 AM Subject: Re: [Asterisk-Users] Polycom SP300 config files http://www.freedomphones.net/polycom/files/ On Sat, 2005-07-09 at 17:03 +1000, Rudolf Ladyzhenskii wrote: Hi, all Sorry for not exactly on-topic question I got Polycom SP300 phones. Somehow they did not come with software. I will call them on Monday, but in the meantime, I would like to get them going. I need Polycom configuration template files (phone.cfg, sip.cfg and whatever else they supply). Did not find them on the Polycom site. Can someone e-mail those to me ([EMAIL PROTECTED])? Then I will be able to work on the weekend. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_bluetooth, no voice
I am running Asterisk 1.0.9 with chan_bluetooth (kernel 2.6.11) I have managed to get the module loaded and it connects to my phone and dials (Nokia 6310i) okay, but once the call connects I hear no sound on either end. It is not who I am calling since it works with any other bluetooth headset or without one. It isn't the SIP phone or codecs because they work with any other number. I think I have the right modules loaded (l2cap, rfcomm, sco, hci_usb, bluetooth, snd_bt_sco), and other bluetooth tools work fine (hcitool scan for example). Using CSR USB adapter (belkin f8t001). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXphone - ip address - extension number.
If you use 2.4, consider 2.6, as its ztdummy works better. If you use 2.6, you may be using udev, and need to read README.udev . Bingo! I had read the README.udev, but had not noticed any make-time udev related messages so chose to ignore its contents. Bad, bad boy - naughty me. Anyway, dumping those lines into the 50-udev-rules file has solved this issue perfectly. I can now modprobe zaptel then ztdummy with no errors and asterisk loads opens all the IAX stuff confirms that it is listening on port 4569. Cool - well spotted. Now to get rid of these darn issues: *CLI Jul 9 10:23:46 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! -- Accepting unauthenticated call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, IAX/z2|20|tr) in new stack Jul 9 10:23:57 WARNING[8102]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 10:23:57 NOTICE[8102]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 10:24:07 WARNING[8102]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/5' Jul 9 10:24:18 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:24:20 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 I'm sure its now possibly a phone setup, but being green at this asterisk stuff, I let them sell me the phones without setup manuals or anything. grrr. In response to the ethereal queries I was asked: All PCs phones on this network plug into the same 3com switch, port 8 of this switch links to a netgear DG834 (my ADSL modem/router/firewall). The three remaining ports on the netgear are unused, but if I plug anything into them, the netgear DHCP offers them a compatible IP and they can see everything plugged into the 3COM. Note that *all* equipment that is permanently connected has a *fixed* IP and does not get one from the netgear DHCP service. In short, the fact that ethereal saw nothing coming from the phones suggest a phone setup issue. However the fact that when I dial *1202 (*1 being the phones IAX dial-prefix), asterisk responds to the request, shows that the phones do however communicate in some recognisable way. Maybe I should also fiddle with a minimum iax.conf and extensions.conf. Now we're getting there. In one of your previous emails, you indicated: 8) IAX username - still left blank 9) IAX password - still left blank Edit those to something valid, don't leave them blank. Then, in iax.conf, enter the same username and password. username= secret= Power cycle the phone and _now_ it should register properly. (That's why you are getting register_verify: Empty registration from 192.168.0.202 in the above CLI. To jump ahead a little, if the phone has a config entry for type of dtmf, set it to rfc2833. This applies more to the sip use then it does to iax, but set it anyway. Also, if the phone's config has a parameter that says something about transmit silence (or words that are something close), be sure to set that to yes. Regarding your comments about the 3com and netgear switches, ethernet switches do not forward all packets to every port. They are smart enough to know where each MAC resides, and only forward packets out a switch port if the packet is destined for the device attached to that port. So, in your ethereal packet traces all you will ever see is broadcast packets (which are sent out all ports). If you need to run ethereal again with those switches, you will have to install and run it on the asterisk box. Otherwise, you will never see the desired traffic. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth, no voice
On Sat, 2005-07-09 at 13:52 +0100, Shaun Orchard wrote: See inline I am running Asterisk 1.0.9 with chan_bluetooth (kernel 2.6.11) 1.0.9 with kernel 2.6.12 I have managed to get the module loaded and it connects to my phone and dials (Nokia 6310i) Nokia 6680 but once the call connects I hear no sound on either end. My asterisk doesn't sense the connection It is not who I am calling since it works with any other bluetooth headset or without one. It isn't the SIP phone or codecs because they work with any other number. I think I have the right modules loaded (l2cap, rfcomm, sco, hci_usb, bluetooth, snd_bt_sco), and other bluetooth tools work fine (hcitool scan for example). Using CSR USB adapter (belkin f8t001). MSI USB adapter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can we register users in oh323.conf ?
Can anyone tell me how to register users in oh323.conf ... i m currently using Netmeeting SJPhone n i can call from/to them without creating user accountsproblem is that my Netphone KU1120(IP phone) uses uid and password for authentication... is there any way to define users like sip.conf or iax.conf ... ?? or anyother way... plz guide me Do you Yahoo!? Make Yahoo! your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can we register users in oh323.conf ?
On Sat, 2005-07-09 at 06:21 -0700, Adeel Ali wrote: Can anyone tell me how to register users in oh323.conf ... i m currently using Netmeeting SJPhone n i can call from/to them without creating user accountsproblem is that my Netphone KU1120(IP phone) uses uid and password for authentication... is there any way to Net2phone uses propietary n2p protocol. define users like sip.conf or iax.conf ... ?? or anyother way... plz guide me __ Do you Yahoo!? Make Yahoo! your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] editing ring time
wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] editing ring time
wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. It kind of looks like maybe the OP's email reader might have a problem. (Either stuck sending the same old thing, or, he's not getting the many responses that have been sent.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Dial Out
Title: Re: [Asterisk-Users] editing ring time I am using the auto-dial-out feature to play recordings. I create the call files, place them in the outgoing directory and off they go. The problem is that the number I am dialing does not get stored in CDR. One suggestion was to put this number in the callerid field. Problem with that is that the recipient will see their own number, which is unacceptable. I must show a toll-free number. I've tried resetting the callerid in thedialplan context before the CDR is stored. That works great, except if the call goes unanswered, it never makes it into the dialplan logic. I must somehow get this number into CDR, as I need it to match back to a customer activity database. Any suggestions? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] editing ring time
Rich Adamson wrote: wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. It kind of looks like maybe the OP's email reader might have a problem. (Either stuck sending the same old thing, or, he's not getting the many responses that have been sent.) I too have noticed that on my thread sometimes my postings take over an hour to pop up (Maybe this list server engine is clever enough to know when someone hogs too much bandwidth ;-) ) zoltan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can we register users in oh323.conf ?
what if a phone is a H323 phone??? On 7/9/05, Guillermo Salas M [EMAIL PROTECTED] wrote: On Sat, 2005-07-09 at 06:21 -0700, Adeel Ali wrote: Can anyone tell me how to register users in oh323.conf ... i m currently using Netmeeting SJPhone n i can call from/to them without creating user accountsproblem is that my Netphone KU1120(IP phone) uses uid and password for authentication... is there any way to Net2phone uses propietary n2p protocol. define users like sip.conf or iax.conf ... ?? or anyother way... plz guide me __ Do you Yahoo!? Make Yahoo! your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] editing ring time
On Saturday 09 July 2005 10:33, Zoltan Szecsei wrote: I too have noticed that on my thread sometimes my postings take over an hour to pop up (Maybe this list server engine is clever enough to know when someone hogs too much bandwidth ;-) ) It's a mailing list, not a realtime interactive chat session. Not seeing an answer in an hour is no reason to post again. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration
Hi again, Well, thanks for the details steps. But before I received your mail I had already installed [EMAIL PROTECTED] v.1.3 and updated it with OH323 add-on. It is a zip file which when you install you get all the libraries installed and compiled for you. Now, one last step for me which I need your help all with. What is needed to get the CCM and Asterisk to exchange calls over H323? I mean which config files needs to be updated. I now have oh323.conf shown and ready. Thanks Walid Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration From: Vamsi Pottangi [EMAIL PROTECTED] Date: Mon, 27 Jun 2005 11:16:49 +0530 Reply-to: Vamsi Pottangi [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] The below worked for me to integrate with CCM. pwlib-v1_6_6 openh323-v1_13_5 asterisk-oh323-0.7.1 The only change I made was -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- Steps to follow: --- To enable H323 for inter-op with Cisco Call Manager (H.323) cp pwlib-v1_6_6-src.tar.gz openh323-v1_13_5-src.tar.gz asterisk-oh323-0.7.1.tar.gz /usr/src/ cd /usr/src tar zxf pwlib-v1_6_6-src.tar.gz tar zxf openh323-v1_13_5-src.tar.gz tar zxf asterisk-oh323-0.7.1.tar.gz - Set Environment variables PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib cd /usr/src/pwlib ./configure make opt cd /usr/src/openh323 ./configure -- Remove the line 433 (:protected) in /usr/src/openh323/include/gkserver.h else you would get the below error during compilation /usr/src/openh323/include/gkserver.h:434: error: `virtual H323Transaction::Response H323GatekeeperRRQ::OnHandlePDU()' is protected -- make opt cd /usr/src/asterisk-oh323-0.7.1 Edit makefile and set the paths/options according to your system. Type "make" to build the oh323wrap library and the ASTERISK OH323 channel driver. - If compiling fails, then change the makefile to reflect the below CPPFLAGS=$(OPENH323FLAGS) -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -I/usr/src/pwlib/include -DPTRACING -I/usr/src/openh323/include -DHAS_OSS -Wall -x c++ -Os --- Type "make install" to install the binaries. This will also install a sample configuration file, if there isn't one. Next, add to your LD_LIBRARY_PATH the path where the oh323wrap library was installed (or edit your /etc/ld.so.conf file, add the library path, and run "ldconfig"). Thanks, ~Vamsi On 6/26/05, Walid Azab [EMAIL PROTECTED] wrote: I have previously tried the Asterisk/OH323/PWLIB/GNUGK combination and had problems compiling OH323. I will try again from a clean installation. On the other hand, can you send me any useful links or guides that you already used. This can make our trial and error efforts much less. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Greg Oliver Sent: Sunday, June 26, 2005 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration We have successfully connect * .9x 1.0.x with CCM 3.3.x and up using both gatekeeper and no gatekeeper.. Using SIP usually with CCM 4.0 and up.. With CCM 3.3.x, there is a limitation where the gateway H323 in your case cannot use IP addresses, so the Asterisk box has to have correct DNS entries to resolbve your asterisk ox.. Then just use regular route patterns and direct it to asterisk.. That works well. You may also want to make sure your compatibility matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more issues than I care to talk about. The GNUGk web site has the best matrix to follow.. Thanks, GReg On Sat, 2005-06-25 at 10:39 -0500, [EMAIL PROTECTED] wrote: Use a gatekeeper and have both boxes register with the gatekeeper. That way you can specify what numbers go where. From everything I have tested, * will NOT register with CCM. When I added in a gatekeeper and had both sides register with it, everything works. Walid Azab wrote: Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
RE: [Asterisk-Users] editing ring time
This is becoming a waste of time and bandwidth. He doesn't know the dial-command, he can't use google and he can't read email... I don't think he'll be around here much longer. I would say by ignoring his posts we're only replying in kind. -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: Saturday, July 09, 2005 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] editing ring time wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXphone - ip address - extension number.
Ha! yes - we are getting there - hopefully soon you will allow yourself some time for anything other than me. see inbetween - and then at the end. Rich Adamson wrote: Now we're getting there. In one of your previous emails, you indicated: 8) IAX username - still left blank 9) IAX password - still left blank Edit those to something valid, don't leave them blank. Then, in iax.conf, enter the same username and password. username= secret= did that earlier today and started getting different errors, so I knew I was on the right track. Power cycle the phone and _now_ it should register properly. (That's why you are getting register_verify: Empty registration from 192.168.0.202 in the above CLI. yep To jump ahead a little, if the phone has a config entry for type of dtmf, set it to rfc2833. This applies more to the sip use then it does to iax, but set it anyway. didnt see it anywhere on the phone web-setup, but I've set this in iax.conf Also, if the phone's config has a parameter that says something about transmit silence (or words that are something close), be sure to set that to yes. couldn't find anything like this Regarding your comments about the 3com and netgear switches, ethernet switches do not forward all packets to every port. They are smart enough to know where each MAC resides, and only forward packets out a switch port if the packet is destined for the device attached to that port. So, in your ethereal packet traces all you will ever see is broadcast packets (which are sent out all ports). If you need to run ethereal again with those switches, you will have to install and run it on the asterisk box. Otherwise, you will never see the desired traffic. I was running ethereal on the same box as asterisk!!! Rich OK, Based on last nights breakthru this mornings fiddling, I have minimised iax.conf filled in everything on the phone itself. Hallelujah! (I'm sure Rich Carlos will agree) :-) I'm still not ringing the other phone, but that is now surely a dialplan issue - extensions.conf has been totally ignored and that can be tomorrows fun as my wife I have a nice dinner date tonight. * iax.conf: *** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 all0w=ulaw all0w=alaw all0w=gsm jitterbuffer=yes [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 [z2] type=friend user=z2 secret=z2 context=geograph host=dynamic dtmfmode=rfx2833 *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Jul 9 16:49:59 WARNING[13788]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 16:49:59 NOTICE[13788]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 16:50:09 WARNING[13788]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/3' -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Accepting AUTHENTICATED call from 192.168.0.202, requested format = 4, actual format = 256 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, IAX/z1|20|tr) in new stack Jul 9 16:51:29 WARNING[13788]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 16:51:29 NOTICE[13788]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 16:51:39 WARNING[13788]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/1' *CLI stop now ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone w/ XML browser
PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). - Dan Pavel Jezek wrote: Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so any other recommendation? Seems, that xml minibrowser isn't obvious even in high end phone, but I think that via this function can be phone very extensible... thanks PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] McLeod Integrated T1 - no PRI?
We have a McLeod T1 and they told us specifically that it was a PRI, ended up being em_wink. Make sure they really have it setup right. -- ~Andy Brezinsky On Friday 08 July 2005 5:45 pm, Kristian Kielhofner wrote: Hello everyone, We have recently turned up a new T1 from McLeod (Midwestern CLEC). It is configured like so: /etc/zaptel.conf: loadzone=us defaultzone=us span=1,0,0,esf,b8zs ;(also tried 1,1,0,esf,b8zs) bchan=13-23 nethdlc=1-12 dchan=24 /etc/zapata.conf: switchtype=national context=pri-in signalling=pri_cpe group=1 channel = 13-23 I can get hdlc0 (and pvc0) up just fine after the appropriate sethdlc and ifconfig commands. Works perfectly. No alarms, everything looks good. PRI, however, refuses to work... pbx*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Evidently, this is a Lucent 5ess switch but I am supposed to use national. I have tried both, and every combination of pri_cpe, pri_net, switchtype, etc. to no avail. The McLeod switch tech says that everything looks fine on their end, and they too verify that HDLC on the T is up and there are no alarms. We are both totally confused. I am pretty sure that it is something on their end, because I have configured many a T1/PRI without problems. If anything, the fact that this is an integrated product may have something to do with it, but I am bamboozled by the fact that HDLC data works and PRI does not. I also tried it without the nethdlc lines, no dice. pri intense debug span 1 shows SABME's going out with nothing else happening. Currently this T is connected to a Te110p, but we also tried a te405p with the same results. Same thing with a Sangoma A101. We even connected them back to back, changed one to pri_net, and were able to bring up the PRI in between two Asterisk servers. Looks like it's McLeod? Does anyone have any ideas? Any magic words to give to the people at McLeod to get this running? Any success/failure stories with McLeod in general? Running CVS HEAD, tried three versions from three separate times, including the most recent from yesterday. Thanks! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users pgp8DCobGCTkZ.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can we register users in oh323.conf ?
On Sat, 2005-07-09 at 22:48 +0800, IM.Nobody wrote: what if a phone is a H323 phone??? You need gnugk to register H.323 phone. You must have to include your SIP extensions in your oh323.conf, in example: ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; Colocar las extensiones SIP en esta seccion alias=asterisk ; For the voice mail alias=*98 ; For the SIP Phones alias=100 alias=101 alias=102 alias=103 alias=104 alias=105 alias=106 alias=107 alias=108 alias=109 alias=110 alias=200 alias=201 alias=202 alias=203 alias=204 alias=205 alias=206 alias=207 alias=208 alias=209 alias=210 alias=500 alias=501 alias=502 On 7/9/05, Guillermo Salas M [EMAIL PROTECTED] wrote: On Sat, 2005-07-09 at 06:21 -0700, Adeel Ali wrote: Can anyone tell me how to register users in oh323.conf ... i m currently using Netmeeting SJPhone n i can call from/to them without creating user accountsproblem is that my Netphone KU1120(IP phone) uses uid and password for authentication... is there any way to Net2phone uses propietary n2p protocol. define users like sip.conf or iax.conf ... ?? or anyother way... plz guide me __ Do you Yahoo!? Make Yahoo! your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] editing ring time
He was wanting to edit the dialtime in astcc. I have sent him a patched copy and I think the issue has been resolved. Darren Wiebe [EMAIL PROTECTED] Jay Milk wrote: This is becoming a waste of time and bandwidth. He doesn't know the dial-command, he can't use google and he can't read email... I don't think he'll be around here much longer. I would say by ignoring his posts we're only replying in kind. -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: Saturday, July 09, 2005 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] editing ring time wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + spandsp
Hello: I dont know, if is my question to do hier, or in the dev-list, but anyway: I 've installed Asterisk (head, development because I need Realtime), but when I try to apply the patch I 've got many errors, reason why I wrote myself the apps/Makefile. (Of course, first, I compiled spandsp, etc.) Then, I try to compile Asterisk, but it 's impossible: The output: app_rxfax.c:14:1: warning: _GNU_SOURCE redefined command line:4:1: warning: this is the location of the previous definition app_rxfax.c: In function `phase_e_handler': app_rxfax.c:70: error: structure has no member named `resolution' app_rxfax.c:77: error: structure has no member named `callerid' app_rxfax.c:81: error: structure has no member named `resolution' app_rxfax.c:83: error: structure has no member named `rx_file' make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Can anyone help me? Thanks Leonardo F. Bauchwitz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910
I have found out that there are 2 modules.conf in [EMAIL PROTECTED]. One in /etc/asterisk/modules.conf and another in /etc/asterisk/default/modules.conf Also skinny.conf is located in /etc/asterisk/default/skinny.conf, however sccp.conf is in /etc/asterisk/sccp.conf Should I copy sccp.conf to --/default? Also show I change modules.conf from --/default as well? I will post what I shows in log of what you said. Thanks Sergio Chersovani [EMAIL PROTECTED] wrote: Javier Chia ha scritto: I did that, the phone logged in, but is unable to make nor recive calls.did you disabled the skinny channel on modules.conf?noload = chan_skinny.soload = chan_sccp.soif this does not work for you, you should post your sccp.conf. It should be a config problemSergio___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Stay connected, organized, and protected. Take the tour___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] editing ring time
Go into the file astcc.agi and find the exec dial line and edit it. - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, July 09, 2005 8:40 AM Subject: Re: [Asterisk-Users] editing ring time He was wanting to edit the dialtime in astcc. I have sent him a patched copy and I think the issue has been resolved. Darren Wiebe [EMAIL PROTECTED] Jay Milk wrote: This is becoming a waste of time and bandwidth. He doesn't know the dial-command, he can't use google and he can't read email... I don't think he'll be around here much longer. I would say by ignoring his posts we're only replying in kind. -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: Saturday, July 09, 2005 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] editing ring time wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + spandsp
On Sat, Jul 09, 2005 at 01:15:16PM +, Leonardo F. Bauchwitz wrote: Hello: I dont know, if is my question to do hier, or in the dev-list, but anyway: I 've installed Asterisk (head, development because I need Realtime), but when I try to apply the patch I 've got many errors, reason why I wrote myself the apps/Makefile. (Of course, first, I compiled spandsp, etc.) Then, I try to compile Asterisk, but it 's impossible: For the record, the debian source package asterisk-apps-spandsp builds out-of-tree just fine. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone w/ XML browser
On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910
Hi, I have uploaded the all the .conf files and screenshots of the log and Xlite. http://www.amsystems.cc/7910/files.zip Please unzip them and check what is wrong. Thanks, JavierSergio Chersovani [EMAIL PROTECTED] wrote: Javier Chia ha scritto:The phone is now logged in but can´t place nor receivecalls. It keeps giving Busy tone when I try to dial anumber. it does happen when there are no matching extensions for the number you are dialinginternal context is ok?you can dial just internal context extensions from the 7910But in: Asterisk*CLI shows the following: please from the cli:sccp debug 10place a call from the cisco 7910place a call from X-lite to the cisco 7910post the log so I can see what is wrongSergio___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent Queue, Silent Calls Problem
I have an issue with silent calls when an agent gets a call from the queue What happens is - The system dials a call (agent call) - The caller picks up - Asterisk sees the person picked up - Transfered to an agent - Agents phone automaticly picks up (sjphone auto accept on) -The user hears nothing says Hello, Hello, Hello ??? - Asterisk sees agent as 'Available' (even though hes on the phone) - Asterisk times out (like it was ringing the phone) -I have set the timeout to different amounts of time and it changes the amount of time that the phone automaticly hangs up - Asterisk times out, sjphone hangs up - Person on the line goes into the queue - Waits the 15 seconds to retry (or whatever it was set to) - Transfered back to the agent, and they are there .. both can hear each other Any ideas of how to fix this problem. I have tried to remove auto answer, and it does the same. I have not tried other phones though. Evan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote SIP Connection using Asterisk // Cisco 7940's
Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, Ross ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk ring a specific extension based on the number the outside caller dialed?
I use the distinctive ring detection for our front door intercom, and I've noticed it's not 100% effective. If this is a business type line, I think I might try to find another solution if it's important that it works 100% of the time. -Mishehu Andrew Kohlsmith wrote: On Friday 08 July 2005 17:01, Jeff Ramsey wrote: I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? Asterisk can detect distinctive ringing, so if your telco does it this way and it's in a format Asterisk accepts, then yes. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:42cf027c19787645211667! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXphone - ip address - extension number.
Based on last nights breakthru this mornings fiddling, I have minimised iax.conf filled in everything on the phone itself. Hallelujah! (I'm sure Rich Carlos will agree) :-) I'm still not ringing the other phone, but that is now surely a dialplan issue - extensions.conf has been totally ignored and that can be tomorrows fun as my wife I have a nice dinner date tonight. * iax.conf: *** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 all0w=ulaw all0w=alaw all0w=gsm Look closely at the above four lines. In the allow statement, that appears to be a zero. Change that to allow. Also, I don't know which codecs the phone supports, but you might start playing with disallow=all allow=ulaw and go from there. [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 If you look at /usr/src/asterisk/configs/iax.conf.sample, you'll find that dtmfmode=rfx2833 is not a valid iax statement. Plus its spelled wrong (its rfc2833). Remove it, but add it into your sip.conf if you're going to play with sip. *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 Here is the key: That is telling you it can't find a compatible codec to allow the call to complete. That's the basis for the comments above about the allow=ulaw. *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Note the above IAX. I think that should be IAX2, so look in your extensions.conf for a dial statement that looks like Dial(IAX/ and change it to Dial(IAX2/. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone w/ XML browser
thank you Brian, but seems, that Polycom phones are not very good option for general corporate use and even not for use with asterisk (* explicitly unsupported!), look: from voipsupply.com: Please Note: Polycom phones are not supported under Asterisk Open Source PBX. from Polycom FAQ: Can the phones be used independent of a Technology Partner’s platform? No. In order to support full business phone features, the SoundPoint IP is required to operate in conjunction with Partners’ IP PBX... Can the phones support LDAP directories? Currently there is no support for directories like LDAP. Is there a web browser built into the phone? Polycom does not currently support this capability. I found avaya phones, that have nice features as I mentioned before (e.g. xml browser) , any experiance with avaya SIP phones and their cost? PJ Brian Roy wrote: On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phantom incomming calls from asterisk
Many telcos do an automated once a day or once a week or ?? line test,
which can appear as an incoming call to some devices.
If you unplug your telco line and the events disappear, perhaps that is
what is happening?
John Novack
John Millican wrote:
About once a day I have noticed a phantom incoming call with a caller ID of
[EMAIL PROTECTED]..cut off. When I answer the call there is a dial tone
and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I have an * box at my house with 1 zap (pstn on a X100p clone from digit
networks) channel and one sip(linksys ATA). I am getting ring on the ATA but
there is no call comming in from the pstn. The following is the CLI output
when this happens. I know that there is no call on the pstn because i have
an "emergency phone"(frequent power outages) still connected to the PSTN
parallel to the * box and it never rings. All the SIP stuff is on an internal
lan only. I only call out on PSTN since all I have available here in
nowheare land is dial up :-( All work flawlessly except for this one
problem.
- Starting simple switch on 'Zap/1-1'
Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2
(Ring/Answered)...
-- Executing Dial("Zap/1-1", "sip/677|35") in new stack
-- Called 677
-- SIP/677-55a8 is ringing
== Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Is there anything for Zap like sip debug? My first guess is that I am getting
some sort of blip in ring voltage on the PSTN but have no way to prove this.
As a posible logic check I unplugged from PSTN, which put zap into Red alarm
of course, and then i get no phantom calls. Is there something in the zap
driver that shuts down when in red alarm?
Any Ideas?
John M
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Re: [Asterisk-Users] SIP phone w/ XML browser
Pavel Jezek wrote: thank you Brian, but seems, that Polycom phones are not very good option for general corporate use and even not for use with asterisk (* explicitly unsupported!), look: from voipsupply.com: Please Note: Polycom phones are not supported under Asterisk Open Source PBX. from Polycom FAQ: Can the phones be used independent of a Technology Partner’s platform? No. In order to support full business phone features, the SoundPoint IP is required to operate in conjunction with Partners’ IP PBX... Can the phones support LDAP directories? Currently there is no support for directories like LDAP. Is there a web browser built into the phone? Polycom does not currently support this capability. I found avaya phones, that have nice features as I mentioned before (e.g. xml browser) , any experiance with avaya SIP phones and their cost? PJ Well, we have over 100 Polycom phones deployed with Asterisk in a corporate environment and they are working extremely well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modifying astcc
Hi:
Astcc is working fine, except for one thing. It
doesn't give the called party enough time to answer
the phone. If nobody picks up in two rings, astcc
reports back no answer and hangs-up. The only instant
NOANSWER value was mentioned in astcc.agi script is:
elsif ($res eq NOANSWER) {
$res =
mystreamfile(astcc-noanswer);
Please help me find what and where to change to
control the time astcc give to the called party to
answer.
Regards;
Chawki Hammoud
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Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910
Javier Chia [EMAIL PROTECTED] : I have uploaded the all the .conf files and screenshots of the log and Xlite. well let's start from extensions.conf ; Cisco 7910 replace [121] with [sccp] because in your sccp.conf the context is sccp exten = 121,1,SetCalledParty(PRUEBA121) exten = 121,2,Dial(PRUEBA/Test1,10,tr) the dial cmd is wrong, this is the correct one (according to your sccp.conf): exten = 121,2,Dial(SCCP/ian,10,tr) exten = 121,3,Voicemail,u121 exten = 121,102,Voicemail,b121 syntax errors on sccp.conf replace [SEP0008E399E223] ] with [SEP0008E399E223] callwaiting = 1 is deprecated, use incominglimit = 1 intercoms are not implemented so you can remove these lines. [intercom] description = Reception Intercom device = SEP0008E399E223 ; device = SEP000AB7567E18 dígame si todo trabaja :-) Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Xten eyeBeam Video Problems
Are you sure that the video is set up correctly? If you have a cheap webcam you have to turn off video hardware acceleration. Cheers. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Blake Krone Sent: Friday, July 08, 2005 5:53 AM To: Matt Riddell Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Xten eyeBeam Video Problems I only have the basic h.263 enabled in Xten. Everytime I start sending video it just shows noise, I can see in the log that it's trying to use the 263 codec. On 7/7/05, Matt Riddell [EMAIL PROTECTED] wrote: Blake Krone wrote: Hello all, I HAD video working before I upgraded to 1.08 (latest stable with Gentoo) and now it won't work. I just see noise bars and not the video. I know the camera works as I can use it in other programs such as AIM Yahoo. Which codec are you using for video in the eyeBeam? We have video IVR, voicemail, billing for video calls etc working fine here with multiple hardware and also the eyeBeam. My recommendation would be to allow only one video codec at a time in eyeBeam's confs. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] editing ring time
Hi: Please save the bandwidth if your answer is going to be go to google or read the wiki. Regards; Chawki --- Jay Milk [EMAIL PROTECTED] wrote: This is becoming a waste of time and bandwidth. He doesn't know the dial-command, he can't use google and he can't read email... I don't think he'll be around here much longer. I would say by ignoring his posts we're only replying in kind. -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: Saturday, July 09, 2005 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] editing ring time wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modifying astcc
If you searched the archives you might find the answer from the past
couple of days. I'll resend it.
In astcc.agi there is are lines similar to this:
$dialstr = IAX2/$res-{path}/$phone|30|HL( . ($maxtime * 60 *
1000) . :6:3);
change the 30 to however many seconds you want.
Darren Wiebe
[EMAIL PROTECTED]
chawki hammoud wrote:
Hi:
Astcc is working fine, except for one thing. It
doesn't give the called party enough time to answer
the phone. If nobody picks up in two rings, astcc
reports back no answer and hangs-up. The only instant
NOANSWER value was mentioned in astcc.agi script is:
elsif ($res eq NOANSWER) {
$res =
mystreamfile(astcc-noanswer);
Please help me find what and where to change to
control the time astcc give to the called party to
answer.
Regards;
Chawki Hammoud
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Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910
You have to change the sip.conf and set context=sccp for x-lite to be able to dial 121 http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Introduction Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Xten eyeBeam Video Problems
It is by no means cheap, and it works doing an echo test, just doesn't work when I try to transmit to another side. On 7/9/05, Storm D. J. Petersen [EMAIL PROTECTED] wrote: Are you sure that the video is set up correctly? If you have a cheap webcam you have to turn off video hardware acceleration. Cheers. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Blake Krone Sent: Friday, July 08, 2005 5:53 AM To: Matt Riddell Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Xten eyeBeam Video Problems I only have the basic h.263 enabled in Xten. Everytime I start sending video it just shows noise, I can see in the log that it's trying to use the 263 codec. On 7/7/05, Matt Riddell [EMAIL PROTECTED] wrote: Blake Krone wrote: Hello all, I HAD video working before I upgraded to 1.08 (latest stable with Gentoo) and now it won't work. I just see noise bars and not the video. I know the camera works as I can use it in other programs such as AIM Yahoo. Which codec are you using for video in the eyeBeam? We have video IVR, voicemail, billing for video calls etc working fine here with multiple hardware and also the eyeBeam. My recommendation would be to allow only one video codec at a time in eyeBeam's confs. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom soundpoint 300 sip phone and hold music
I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] polycom soundpoint 300 sip phone and hold music
I think that it might have to do with the codec that is being used. I had a problem trying to get the hold music to work with calls that went over our trunk. I can't remember which one did not work but hopefully this will give you a direction. Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derrick Stensrud Sent: Saturday, July 09, 2005 3:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] polycom soundpoint 300 sip phone and hold music I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 version 0.6.6.
Hi, I have downloaded asterisk-oh323-0.6.6.tar pwlib-Janus_patch4-src-tar openh323-Janus_patch4-src-tar pwlib and openh323 compiled fine as instructed. When I tried to compile asterisk-oh323 I am getting this and anybody know howto fix this? [EMAIL PROTECTED] oh323]# cd asterisk-oh323-0.6.6 [EMAIL PROTECTED] asterisk-oh323-0.6.6]# ls asterisk-driver CONFIGURATION Makefile rpmTESTS BUGS COPYINGREADMErules.mak wrapper [EMAIL PROTECTED] asterisk-oh323-0.6.6]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/root/oh323/asterisk-oh323-0.6.6/wrapper' ./check_ver /root/oh323/pwlib pwlib ./check_ver /root/oh323/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/root/oh323/asterisk-oh323-0.6.6/wrapper' make[1]: Entering directory `/root/oh323/asterisk-oh323-0.6.6/asterisk-driver' gcc -Wall -DUSE_OLD_CAPABILITIES_API=1 -march=i686 -pipe -Wstrict-prototypes - Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -fPIC - I/usr/include/asterisk -I../wrapper -c -o chan_oh323.o chan_oh323.c In file included from /usr/include/string.h:33, from chan_oh323.c:34: /usr/local/lib/gcc-lib/i686-pc-linux-gnu/3.3.6/include/stddef.h:213: error: syntax error before typedef In file included from chan_oh323.c:34: /usr/include/string.h:38: error: syntax error before extern /usr/include/string.h:39: error: syntax error before __THROW /usr/include/string.h:43: error: syntax error before __THROW /usr/include/string.h:56: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:58: error: syntax error before extern /usr/include/string.h:58: error: syntax error before __THROW /usr/include/string.h:62: error: syntax error before __THROW /usr/include/string.h:66: error: syntax error before __THROW /usr/include/string.h:80: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:82: error: syntax error before extern /usr/include/string.h:83: error: syntax error before __THROW /usr/include/string.h:86: error: syntax error before __THROW /usr/include/string.h:90: error: syntax error before __THROW /usr/include/string.h:93: error: syntax error before __THROW /usr/include/string.h:97: error: syntax error before __THROW /usr/include/string.h:100: error: syntax error before __THROW /usr/include/string.h:104: error: syntax error before __THROW /usr/include/string.h:107: error: syntax error before __THROW /usr/include/string.h:160: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:162: error: syntax error before extern /usr/include/string.h:162: error: syntax error before __THROW /usr/include/string.h:164: error: syntax error before __THROW /usr/include/string.h:173: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:176: error: syntax error before extern /usr/include/string.h:177: error: syntax error before __THROW /usr/include/string.h:181: error: syntax error before __THROW /usr/include/string.h:184: error: syntax error before __THROW /usr/include/string.h:187: error: syntax error before __THROW /usr/include/string.h:192: error: syntax error before __THROW /usr/include/string.h:197: error: syntax error before extern /usr/include/string.h:199: error: syntax error before __THROW /usr/include/string.h:230: error: syntax error before extern ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXphone - ip address - extension number.
Zoltan, If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. Regarding Ethereal, you should do tcpdump -I ethx (x=number of the port) -W filename, when you finish close it with Ctrl-z, and then you can see the file on the Asterisk or move it to another computer with Etherreal and open it (That is the way I do, so I see what Asterisk gets). Have a great weekend. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Saturday, July 09, 2005 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Ha! yes - we are getting there - hopefully soon you will allow yourself some time for anything other than me. see inbetween - and then at the end. Rich Adamson wrote: Now we're getting there. In one of your previous emails, you indicated: 8) IAX username - still left blank 9) IAX password - still left blank Edit those to something valid, don't leave them blank. Then, in iax.conf, enter the same username and password. username= secret= did that earlier today and started getting different errors, so I knew I was on the right track. Power cycle the phone and _now_ it should register properly. (That's why you are getting register_verify: Empty registration from 192.168.0.202 in the above CLI. yep To jump ahead a little, if the phone has a config entry for type of dtmf, set it to rfc2833. This applies more to the sip use then it does to iax, but set it anyway. didnt see it anywhere on the phone web-setup, but I've set this in iax.conf Also, if the phone's config has a parameter that says something about transmit silence (or words that are something close), be sure to set that to yes. couldn't find anything like this Regarding your comments about the 3com and netgear switches, ethernet switches do not forward all packets to every port. They are smart enough to know where each MAC resides, and only forward packets out a switch port if the packet is destined for the device attached to that port. So, in your ethereal packet traces all you will ever see is broadcast packets (which are sent out all ports). If you need to run ethereal again with those switches, you will have to install and run it on the asterisk box. Otherwise, you will never see the desired traffic. I was running ethereal on the same box as asterisk!!! Rich OK, Based on last nights breakthru this mornings fiddling, I have minimised iax.conf filled in everything on the phone itself. Hallelujah! (I'm sure Rich Carlos will agree) :-) I'm still not ringing the other phone, but that is now surely a dialplan issue - extensions.conf has been totally ignored and that can be tomorrows fun as my wife I have a nice dinner date tonight. * iax.conf: *** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 all0w=ulaw all0w=alaw all0w=gsm jitterbuffer=yes [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 [z2] type=friend user=z2 secret=z2 context=geograph host=dynamic dtmfmode=rfx2833 *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Jul 9 16:49:59 WARNING[13788]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 16:49:59 NOTICE[13788]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 16:50:09 WARNING[13788]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/3' -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Accepting AUTHENTICATED call from 192.168.0.202,
Re: [Asterisk-Users] oh323 version 0.6.6.
On Sat, 2005-07-09 at 17:59 -0500, CM Rahman Jr. wrote: Hi, I have downloaded asterisk-oh323-0.6.6.tar pwlib-Janus_patch4-src-tar openh323-Janus_patch4-src-tar pwlib and openh323 compiled fine as instructed. When I tried to compile asterisk-oh323 Try this link: http://www.oinko.net/astrecipes/index.php?n=40 I am getting this and anybody know howto fix this? [EMAIL PROTECTED] oh323]# cd asterisk-oh323-0.6.6 [EMAIL PROTECTED] asterisk-oh323-0.6.6]# ls asterisk-driver CONFIGURATION Makefile rpmTESTS BUGS COPYINGREADMErules.mak wrapper [EMAIL PROTECTED] asterisk-oh323-0.6.6]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/root/oh323/asterisk-oh323-0.6.6/wrapper' ./check_ver /root/oh323/pwlib pwlib ./check_ver /root/oh323/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/root/oh323/asterisk-oh323-0.6.6/wrapper' make[1]: Entering directory `/root/oh323/asterisk-oh323-0.6.6/asterisk-driver' gcc -Wall -DUSE_OLD_CAPABILITIES_API=1 -march=i686 -pipe -Wstrict-prototypes - Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -fPIC - I/usr/include/asterisk -I../wrapper -c -o chan_oh323.o chan_oh323.c In file included from /usr/include/string.h:33, from chan_oh323.c:34: /usr/local/lib/gcc-lib/i686-pc-linux-gnu/3.3.6/include/stddef.h:213: error: syntax error before typedef In file included from chan_oh323.c:34: /usr/include/string.h:38: error: syntax error before extern /usr/include/string.h:39: error: syntax error before __THROW /usr/include/string.h:43: error: syntax error before __THROW /usr/include/string.h:56: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:58: error: syntax error before extern /usr/include/string.h:58: error: syntax error before __THROW /usr/include/string.h:62: error: syntax error before __THROW /usr/include/string.h:66: error: syntax error before __THROW /usr/include/string.h:80: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:82: error: syntax error before extern /usr/include/string.h:83: error: syntax error before __THROW /usr/include/string.h:86: error: syntax error before __THROW /usr/include/string.h:90: error: syntax error before __THROW /usr/include/string.h:93: error: syntax error before __THROW /usr/include/string.h:97: error: syntax error before __THROW /usr/include/string.h:100: error: syntax error before __THROW /usr/include/string.h:104: error: syntax error before __THROW /usr/include/string.h:107: error: syntax error before __THROW /usr/include/string.h:160: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:162: error: syntax error before extern /usr/include/string.h:162: error: syntax error before __THROW /usr/include/string.h:164: error: syntax error before __THROW /usr/include/string.h:173: error: syntax error before __BEGIN_NAMESPACE_STD /usr/include/string.h:176: error: syntax error before extern /usr/include/string.h:177: error: syntax error before __THROW /usr/include/string.h:181: error: syntax error before __THROW /usr/include/string.h:184: error: syntax error before __THROW /usr/include/string.h:187: error: syntax error before __THROW /usr/include/string.h:192: error: syntax error before __THROW /usr/include/string.h:197: error: syntax error before extern /usr/include/string.h:199: error: syntax error before __THROW /usr/include/string.h:230: error: syntax error before extern ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's
Some way you should have a udp filter between you box and your phones. I see that before. Can you call those phones? Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross Overstreet Sent: Saturday, July 09, 2005 2:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, Ross ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's
Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross -- Original Message -- From: Carlos Alperin [EMAIL PROTECTED] Date: Sat, 9 Jul 2005 19:33:31 -0400 Some way you should have a udp filter between you box and your phones. I see that before. Can you call those phones? Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross Overstreet Sent: Saturday, July 09, 2005 2:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, Ross ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's
sounds like the inbound (out from the phone, into the local net) RTP packets are getting dropped.. just a guess here.. whats the output of iptables -L -v Ross Overstreet wrote: Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross -- Original Message -- From: Carlos Alperin [EMAIL PROTECTED] Date: Sat, 9 Jul 2005 19:33:31 -0400 Some way you should have a udp filter between you box and your phones. I see that before. Can you call those phones? Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross Overstreet Sent: Saturday, July 09, 2005 2:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP - no NAT or firewalls. When I take the phones to a remote location (again, public IP - no NAT or firewalls that I know of) the outgoing audio does not work. I can hear the other party, my phones ring, I can dial out, etc, but the other party cannot hear me (even if I dial #'s, etc). Any ideas? Thanks, Ross ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] editing ring time
Your post is just as superfluous as mine... or this very post. But if you scroll down a bit, you'll see a I gave a proper reply yesterday. However, the OP doesn't seem to grasp some basic internet principles, such as... waiting for a response before re-posting, reading a response when it occurs, or checking google or the wiki before posting in the first place. When I see cases like that, and I'm in a good mood and know the answer, I'll post that answer along with a reminder to check google. If I'm in a bad mood, I don't respond. -Original Message- From: chawki hammoud [mailto:[EMAIL PROTECTED] Sent: Saturday, July 09, 2005 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] editing ring time Hi: Please save the bandwidth if your answer is going to be go to google or read the wiki. Regards; Chawki --- Jay Milk [EMAIL PROTECTED] wrote: This is becoming a waste of time and bandwidth. He doesn't know the dial-command, he can't use google and he can't read email... I don't think he'll be around here much longer. I would say by ignoring his posts we're only replying in kind. -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: Saturday, July 09, 2005 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] editing ring time wassim darwish wrote: how to edit the time of ring 3ms to 4ms in astcc since it displays this on console Nobody picked up in 3 ms when nobody picked up the phone in 3ms and then it hangup. please help i have been asking this question from long time and no body answered me yet. Wrong. I saw at least two people answering you. They referred you to the Dial command docs in the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sell on Yahoo! Auctions - no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme recordings
I have a conference set up through MeetMe and I can record each call coming in with the Monitor command. What I would like to move away from is having to then generate multiple files for the final output of these calls. On voip-info.org, there is an 'r' option to record the conference. This does not work on my 1.0.7 version of Asterisk. I looked through the app_meetme.c file and the option is not there either. As a reference, here is a link to the page on voip-info.org that I am refering to: http://voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe I have also setup a separate extension to dial intoin an attempt to record all of the members from one source. What I have found is that the first monitor session records all subsequent members of the conference. For example: Three members log in Member one records all members Member two records two and three Member three records member three I guess my question is what happened to the 'r' recording option in meetme? Thanks, Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + spandsp
Hello Tzafrir: Tzafrir Cohen wrote: On Sat, Jul 09, 2005 at 01:15:16PM +, Leonardo F. Bauchwitz wrote: Hello: I dont know, if is my question to do hier, or in the dev-list, but anyway: I 've installed Asterisk (head, development because I need Realtime), but when I try to apply the patch I 've got many errors, reason why I wrote myself the apps/Makefile. (Of course, first, I compiled spandsp, etc.) Then, I try to compile Asterisk, but it 's impossible: For the record, the debian source package asterisk-apps-spandsp builds out-of-tree just fine. I use Debian and Ututo-e (and I have proved Xorcom :)), but this package, -asterisk-apps-spandsp- support Asterisk Real Time? Now, I work with the development version of Asterisk because support that issue. Leonardo Federico Bauchwitz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FS: Digium TDM04B (PCI with four FXO daughterboards)
Never used. $250 + shipping (your choice of method). - a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM04B Outbound calls
I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2.There is any work around or different setting to avoid this situation? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
