RE: [Asterisk-Users] Agent Cleanup Time?
Title: Message Use the agents.conf file. I believe the option you are looking for is wrapuptime and the parameter is in milliseconds so to have an agent clean up time of say thirty seconds you would add the following line: wrapuptime=3 This parameter can either be specified at different areas of the file to apply to all or just some of the agents. Hope this helps, Robert Jackson -Original Message-From: Jeff Crews [mailto:[EMAIL PROTECTED] Sent: Friday, April 16, 2004 5:49 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Agent Cleanup Time?Previously there was discussion about people seeking the ability to have an delay between calls to the agents so that the agent could "clean-up" or wrap up the documentation on the call that just hung up...before the next call is connected to the agent.Was that option added to Asterisk? And if so...what is it officially called and how do we enable it? Jeff
[Asterisk-Users] VoIP SIP SoftPhone Recommendations
What SoftPhone working very well with *? S.O. is Debian Linux Thanks for your comments. JRR _ MSN Amor: busca tu ½ naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tor2 driver panics with 2 sticks of memory
it looks like some other usb module tries to get loaded and that's what causing it. try to insmod the zaptel & tor2 & run ztcfg -vv instead. or rmmod all the uhci modules... regards Martin On Fri, 16 Apr 2004, Jim Gottlieb wrote: > We use dual Athlon machines with up to three T400P 4-span T1 cards. > > If I have more than one stick of memory (2 1GB modules or 2 512K modules, each > identical), I'm getting a panic soon after I modprobe the tor2 driver. I just > loaded the latest from CVS and I'm still getting the panics, which look in part > like: > > Apr 16 14:42:28 test71 kernel: wait_on_irq, CPU 0: > Apr 16 14:42:44 test71 kernel: irq: 1 [ 0 1 ] > Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] > Apr 16 14:42:47 test71 kernel: Stack dumps: > Apr 16 14:42:47 test71 kernel: CPU 1: 000 > 0 > Apr 16 14:42:47 test71 kernel: 00 > 00 > Apr 16 14:42:47 test71 kernel: 00 > 00 > Apr 16 14:42:47 test71 kernel: Call Trace: [] ohci_hcd_list [usb-ohci] > 0x0 > Apr 16 14:42:47 test71 kernel: [] ohci_hcd_list [usb-ohci] 0x0 > Apr 16 14:42:47 test71 kernel: [] rh_int_timer_do [usb-ohci] 0x0 > Apr 16 14:42:47 test71 kernel: > Apr 16 14:42:47 test71 kernel: > Apr 16 14:42:47 test71 kernel: CPU 0:f6a2bea4 c023f901 0001 fff > f c010a362 c023f916 > Apr 16 14:42:47 test71 kernel: f79ce6a4 f6a2bef8 c017f574 04 > 00 0005 04bf 8a31 > Apr 16 14:42:47 test71 kernel:7f1c0300 01000415 1a131100 170f1200 00 > 00 f6a2a000 f782d978 f782d978 > Apr 16 14:42:47 test71 kernel: Call Trace: [] __global_cli [kernel] 0x > e2 > Apr 16 14:42:47 test71 kernel: [] change_termios [kernel] 0x24 > Apr 16 14:42:47 test71 kernel: [] set_termios [kernel] 0x164 > Apr 16 14:42:47 test71 kernel: [] tty_ioctl [kernel] 0x352 > Apr 16 14:42:47 test71 kernel: [] sys_ioctl [kernel] 0x257 > Apr 16 14:42:47 test71 kernel: [] system_call [kernel] 0x33 > Apr 16 14:42:47 test71 kernel: > Apr 16 14:42:47 test71 last message repeated 2 times > Apr 16 14:42:47 test71 kernel: wait_on_irq, CPU 0: > Apr 16 14:42:47 test71 kernel: irq: 1 [ 0 1 ] > Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] > Apr 16 14:42:47 test71 kernel: Stack dumps: > Apr 16 14:42:47 test71 kernel: CPU 1: 42029098 000 > [...] > > > Any ideas? Thanks... > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX firmware for snom 200s?
We had this discussion recently on this mailing list and the answer is no, not yet. We try to optimize the Asterisk interoperability on SIP level. We at snom can currently not afford to open another development branch. Christian > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Justin Carlson > Sent: Friday, April 16, 2004 9:27 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] IAX firmware for snom 200s? > > is there a firmware for IAX for the snom 200's. or are there any other > hard > phones that use iax(2)? > > Thanks in advance! > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Newbie) help please?
Hi,
i have the same problem in the last week with i4l on linux.
The solution is, that you modify the Dial string in the extension.conf:
from
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
to
Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
I hope, this helps.
Regards,
Andreas.
--On Freitag, April 16, 2004 23:58:20 +0200 Mark Elkins <[EMAIL PROTECTED]>
wrote:
What I've got...
Software:
Linux: Slackware 9.1
Asterisk: out of CVS - so its new.
isdn4k-utils: to test the ISDN Card
Hardware:
PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM
1 x ISDN BRI Card - DIVA EICON (Installed + working)
2 x Grandstream (Barbie?) BT100 SIP Phones.
What Works..
I can call from one phone to the other... get read voicemail...
I can dial from a PSTN phone the BRI Number - and get the * demo
messages
Whats been read..
Lots.. Andy's Getting Started (www.automated.it/guidetoasterisk.htm)
and lots from http://www.voip-info.org/wiki-Asterisk+ISDN4Linux and
I've followed almost every link from www.asterisk.org...
All examples seem to include Digiums hardware :-(
I'm looking for clean, clear examples with a generic ISDN card - which
is my trunk line, and the two SIP phones.
The numbering plan in South Africa is pretty simple
7 digits for local calls
12 digits for long distance
Anyone in S.A. got some example configs to share with?
Currently - I'm stuck with the message..
-- Executing Dial("SIP/phone1-082a", "Modem/g1/8070590") in new stack
Apr 17 00:09:00 WARNING[507919]: chan_modem.c:181 modem_call:
Destination g1/8070590 requres a real destination (device:destination)
-- Couldn't call g1/8070590
-- Hungup 'Modem[i4l]/ttyI1'
... when I dial '98070590' (9 for outside - which I'll make '0' one
day!)
(its late, head hurts, wife is loosing patience)
help? hints?
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
--
"Ich denke, man hat kein Recht, andere zu kontrollieren oder Ihnen etwas
aufzuzwingen, den eigenen Glauben oder die eigene Art zu leben."
- Dalai Lama "Begegnungen".
---
Andreas Czerniak <[EMAIL PROTECTED]> - Kiel - FRG - Fax:+49-431-2000447
PGPkey: http://wwwkeys.nl.pgp.net:11371/pks/lookup?op=get&search=0xEDB224EC
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Newbie) help please?
What I've got...
Software:
Linux: Slackware 9.1
Asterisk: out of CVS - so its new.
isdn4k-utils: to test the ISDN Card
Hardware:
PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM
1 x ISDN BRI Card - DIVA EICON (Installed + working)
2 x Grandstream (Barbie?) BT100 SIP Phones.
What Works..
I can call from one phone to the other... get read voicemail...
I can dial from a PSTN phone the BRI Number - and get the * demo
messages
Whats been read..
Lots.. Andy's Getting Started (www.automated.it/guidetoasterisk.htm)
and lots from http://www.voip-info.org/wiki-Asterisk+ISDN4Linux and
I've followed almost every link from www.asterisk.org...
All examples seem to include Digiums hardware :-(
I'm looking for clean, clear examples with a generic ISDN card - which
is my trunk line, and the two SIP phones.
The numbering plan in South Africa is pretty simple
7 digits for local calls
12 digits for long distance
Anyone in S.A. got some example configs to share with?
Currently - I'm stuck with the message..
-- Executing Dial("SIP/phone1-082a", "Modem/g1/8070590") in new stack
Apr 17 00:09:00 WARNING[507919]: chan_modem.c:181 modem_call:
Destination g1/8070590 requres a real destination (device:destination)
-- Couldn't call g1/8070590
-- Hungup 'Modem[i4l]/ttyI1'
... when I dial '98070590' (9 for outside - which I'll make '0' one
day!)
(its late, head hurts, wife is loosing patience)
help? hints?
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] DLINK DPH-70 with asterisk
> Hello everybody, > > i have DLINK DPH-70 Phone, > > does anybody know if it works with asterisk ? RESPONSE: I bought 2 of these and just got an RMA to return them to D-Link because the would not work properly with any major service provider they were tested with. The SIP firmware that comes with the units is not fully SIP compliant and is designed to work with a few SIP providers in India. Best regards...Bill > > i have g729 codec installed on my asterisk server which the phone supports, and > it gets authenticate with asterisk but when i make a call it says maximum tries > reaches for dialing.. > > regards. > -neo > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tor2 driver panics with 2 sticks of memory
We use dual Athlon machines with up to three T400P 4-span T1 cards. If I have more than one stick of memory (2 1GB modules or 2 512K modules, each identical), I'm getting a panic soon after I modprobe the tor2 driver. I just loaded the latest from CVS and I'm still getting the panics, which look in part like: Apr 16 14:42:28 test71 kernel: wait_on_irq, CPU 0: Apr 16 14:42:44 test71 kernel: irq: 1 [ 0 1 ] Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] Apr 16 14:42:47 test71 kernel: Stack dumps: Apr 16 14:42:47 test71 kernel: CPU 1: 000 0 Apr 16 14:42:47 test71 kernel: 00 00 Apr 16 14:42:47 test71 kernel: 00 00 Apr 16 14:42:47 test71 kernel: Call Trace: [] ohci_hcd_list [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: [] ohci_hcd_list [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: [] rh_int_timer_do [usb-ohci] 0x0 Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 kernel: CPU 0:f6a2bea4 c023f901 0001 fff f c010a362 c023f916 Apr 16 14:42:47 test71 kernel: f79ce6a4 f6a2bef8 c017f574 04 00 0005 04bf 8a31 Apr 16 14:42:47 test71 kernel:7f1c0300 01000415 1a131100 170f1200 00 00 f6a2a000 f782d978 f782d978 Apr 16 14:42:47 test71 kernel: Call Trace: [] __global_cli [kernel] 0x e2 Apr 16 14:42:47 test71 kernel: [] change_termios [kernel] 0x24 Apr 16 14:42:47 test71 kernel: [] set_termios [kernel] 0x164 Apr 16 14:42:47 test71 kernel: [] tty_ioctl [kernel] 0x352 Apr 16 14:42:47 test71 kernel: [] sys_ioctl [kernel] 0x257 Apr 16 14:42:47 test71 kernel: [] system_call [kernel] 0x33 Apr 16 14:42:47 test71 kernel: Apr 16 14:42:47 test71 last message repeated 2 times Apr 16 14:42:47 test71 kernel: wait_on_irq, CPU 0: Apr 16 14:42:47 test71 kernel: irq: 1 [ 0 1 ] Apr 16 14:42:47 test71 kernel: bh: 0 [ 0 0 ] Apr 16 14:42:47 test71 kernel: Stack dumps: Apr 16 14:42:47 test71 kernel: CPU 1: 42029098 000 [...] Any ideas? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent Cleanup Time?
Previously there was discussion about people seeking the ability to have an delay between calls to the agents so that the agent could "clean-up" or wrap up the documentation on the call that just hung up...before the next call is connected to the agent. Was that option added to Asterisk? And if so...what is it officially called and how do we enable it? Jeff
Re: [Asterisk-Users] Warning from Asterisk
[EMAIL PROTECTED] wrote: I get this warning from Asterisk and I want to assess whether it is important, and if so, if I should complain to the telephone manifacturer or start up my programmer's editor: chan_sip.c:5152 handle_response: Host '172.31.1.7' does not implement 'NOTIFY' What does this mean and am I missing some important feature? Asterisk sends a NOTIFY to tell the phone that you have voicemail. The phone is telling Asterisk that it doesn't have a clue of what this kind of SIP message is and what it possibly could do with the following bits of data. So you are not getting the message... Turn on SIP debug in the CLI (as Chris suggested) and you'll see the packets. If you don't want to get that message, remove the mailbox= from the peer configuration in sip.conf Sorry for not answering earlier. You didn't say SIP in the subject, so I didn't react... (hint,hint) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer through AGI
Hi, I have the 4 port TDM card. One small part of my application is to transfer a call. I used the stdexten macro in extensions.conf to define device extension 301 as Zap/1, 302 as Zap/2 etc.. I am using the outgoing directory to place a call on Zap/1 (I cant use 301 as I hear DTMF digits of 301 when I try that) So I use the following. Channel: Zap/1 Application: AGI Data: smvoice Setvar: something So when extenion 301 rings and is answered I hear my application say please hold for XYZ. I then try to execute "EXEC Transfer 302" as an AGI call and I get dial tone or a busy tone. It does not do the transfer. Any ideas or suggestions as to what I am doing wrong and why it will not transfer. Thanks Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DLINK DPH-70 with asterisk
Hello everybody, i have DLINK DPH-70 Phone, does anybody know if it works with asterisk ? i have g729 codec installed on my asterisk server which the phone supports, and it gets authenticate with asterisk but when i make a call it says maximum tries reaches for dialing.. regards. -neo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 going down
I have an * system backending a Fujitsu 9600 system. The interface is a P2P T1 circuit using E&M wink. At 6PM every evening the T1 circuit goes down. I don't have any cron or 'at' jobs that fire at this time. Does anyone have any insight? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SoundStation IP 3000 conference phone on *?
Does anyone have a Polycom SoundStation IP 3000 conference phone working on an * server? Or the Cisco or 3com version? I am looking for a high quality conference table phone that is compatible. Any problems? Many thanks, Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Operator Panel new version and Mailing List
Hi All, Version .04 of the Flash Operator Panel is now available. Someone donated a domain name for the project (thanks turcko!), so it is now available on http://www.asternic.org I have set up a mailing list for the application. So please post your comments, suggestions, bug reports and problems there and not in Asterisk-Users. You can subscribe to the mailing list sending an empty email to [EMAIL PROTECTED] The new version has configurable buttons. You can have more that a hundred buttons on the screen. Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard web browser with Flash plugin. You can see at a glance: * What extensions are busy, ringing or available * Who is talking and to whom (clid, context, priority) * SIP registration status and reachability * Number of users waiting on Queues You can perform these actions: * Hang-up a channel * Transfer a call leg via drag&drop Best regards, -- Nicolas Gudino <[EMAIL PROTECTED]> House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote: I will try disallow=all, thanks, Nat is off. Sip.conf below. If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! It is also happening over IAX with the Cisco phones. I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress. Anything internal is perfect. The CAPI works fine. Its just the audio from the other end. Every now and then I can hear a quick bit of sound. One in 20 calls may work. [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to allow=ulaw allow=alaw tos=lowdelay [20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid="Cisco Phone" <20> accountcode=20 qualify=yes context=sip Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio Craig Waddington wrote: When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just to be clear, you need at least the following (or at least I did): sip.conf: nat=yes reinvite=no SIPDefault.conf (in your tftp directory) nat_enable="0" -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 no audio - sip debug
This is a call coming in through the ISDN to 7940's. Answering with non-codec capability 1 - Is that the problem? SIP Debugging Enabled We're at 10.1.0.11 port 18406 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 <<- 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" ;tag=as03605c88 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> c=IN IP4 10.1.0.11 t=0 0 m=audio 18406 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.119:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" ;tag=as03605c88 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: Content-Length: 0 9 headers, 0 lines pbx01*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" ;tag=as03605c88 To: ;tag=000e3857223c0238011930bc-566c64f8 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: Content-Length: 0 9 headers, 0 lines We're at 10.1.0.11 port 18198 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9 From: "asterisk" ;tag=as286f917d To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> c=IN IP4 10.1.0.11 t=0 0 m=audio 18198 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.120:5060 pbx01*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9 From: "asterisk" ;tag=as286f917d To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9 From: "asterisk" ;tag=as286f917d To: ;tag=000f23ad6e25021c1c1f7e2d-532b7f03 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: Content-Length: 0 9 headers, 0 lines We're at 10.1.0.11 port 29654 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced From: "asterisk" ;tag=as19596a6b To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> sip debug c=IN IP4 10.1.0.11 t=0 0 m=audio 29654 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.125:5060 pbx01*CLI> sip debug Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced From: "asterisk" ;tag=as19596a6b To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: Content-Length: 0 9 headers, 0 lines pbx01*CLI> sip debug Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced From: "asterisk" ;tag=as19596a6b To: ;tag=000f23ac489f00c519541b4d-016de7d7 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: CSCO/6 Contact: Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tracy R Reed Sent: 16 April 2004 19:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: > When we receive or make a call to the outside - they can hear us, but we > cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem to be the case for me. For example: SIP (grandstream, snom) ->Asterisk->NAT->Asterisk->SIP (grandstre
[Asterisk-Users] Dealing with LECs (was Strange T1 Problem)
Matt advised: > And > if you absolutely have to have dynamic CallerID transmission you should > verify that your new carrier will let you do that before you sign a > contract(Make sure you verify it by talking to an actual switch tech from > the carrier, sales people will lie through their teeth to get you to sign > that contract). Good advice and here's more... Once you have the actual switch tech to confirm you can run dynamic CallerID (and specify if the CallerIDs you will be specifying are within the DIDs allocated to your PRI or not), then get it in writing. Best is if they will email you confirmation. Failing that, send them email saying "Further to our telephone conversation of xxyyzz, this is to confirm that with the PRI we will be ordering from aabbcc company we will be able to dynamically specify the CLID on outbound calls and the CLID may or may not be one of our DIDs." >From experience -- tech verbally confirmed that we could do this (we did not get it in writing) and then denied it after we signed the contract. The CLEC will remain unnamed. We are still fighting this so the story is not over but be warned. The sales reps aren't the only ones lying. g. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application
me too [EMAIL PROTECTED] pat munis wrote: interested ... please send me some info. - Original Message - From: "Kyle Hagan" <[EMAIL PROTECTED]> Date: Thu, 15 Apr 2004 09:20:01 -0700 To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] PC based Switchboard application Im interested can you send information? Kyle [EMAIL PROTECTED] - Original Message - From: "Pertti Pikkarainen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, April 10, 2004 2:26 AM Subject: Re: [Asterisk-Users] PC based Switchboard application We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: > When we receive or make a call to the outside - they can hear us, but we > cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem to be the case for me. For example: SIP (grandstream, snom) ->Asterisk->NAT->Asterisk->SIP (grandstream, snom) He can hear me but I can't hear him. In another case I had: IAXclient (soft phone)->NAT->Asterisk->Snom And I could hear him but he could not hear me. Same phone system and settings as above. However as soon as I switched the first users phone to talk directly to my Asterisk box with SIP it worked perfectly. And when I switched the user in the second case to a SIP based soft phone it also worked just fine. SIP has worked better through NAT than IAX (with nat=yes in sip.conf) which is bizarre and contrary to what I have read where IAX should be NAT-safe and SIP not. I have dreams of a world fully converted to IPv6 where NAT no longer exists. Alas, it is but a dream. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Interrupting Dial / Qwest-like transfers
On Fri, 16 Apr 2004, Andrew Kohlsmith wrote: > > 2) While I'm at it...has anyone called Qwest lately? When they transfer > > you, they can stay on the line the entire time, listen with you while it > > rings, then when it says "Press 1 for XXX", then they can dial it for you, > > talk to whoever answers and say "I've got Ryan on the line and needs > > [whatever]", then leave the conversation and let you talk to the new > > person. This whole time you're able to sit and listen to what's going on or > > chat with the first person that's doing all this work for you. Is this > > possible with Asterisk?! > > It's called assisted transfer and no, asterisk currently cannot do that. This is sometimes just a fancy form of a two party conference. You just conference in an outside line, and start dialing. As long as you can hang up on the conference, and leave the other party connected, you are ok. This is sometimes called tandem dialing. > -A. Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 no audio
Yes MOH etc work fine for the receiving end, dialing from outside. I have run X-lite and GS phones on the network on a test machine before this one, and it worked great. Though I haven't had a chance to see if they work or not. I will definatley check my Firewall logs, that's a good point, but the sipura works. It seems codec to me, but I have tried many different confs in sip. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: 16 April 2004 18:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Apr 16, 2004, at 10:04 AM, Craig Waddington wrote: > When we receive or make a call to the outside - they can hear us, but > we cant hear them. With SIP, missing audio is *usually* either a firewall or NAT issue. Check firewall logs and make sure that you aren't seeing packets being lost. Do you have more then one 7940? If so, can they call each other? Also, when people call into your system, do they get audio from asterisk? Does voicemail work? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 no audio
I will try disallow=all, thanks, Nat is off. Sip.conf below. If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! It is also happening over IAX with the Cisco phones. I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress. Anything internal is perfect. The CAPI works fine. Its just the audio from the other end. Every now and then I can hear a quick bit of sound. One in 20 calls may work. [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to allow=ulaw allow=alaw tos=lowdelay [20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid="Cisco Phone" <20> accountcode=20 qualify=yes context=sip Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio Craig Waddington wrote: > When we receive or make a call to the outside - they can hear us, but > we cant hear them. > > It may work 1 of 20 times. I have set canreinvite=no and looked at > many sites but cannot track down this problem. > > Current setup: > > Isdn Eicon Diva card / Capi -> Asterisk à network. > > I have tried adjusting the RTP port in rtp.conf with the Cisco default > ports, no luck. > > Anyone had this problem, and has a fix? > > Thanks. > Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote: When we receive or make a call to the outside – they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks. Make sure you don't have the Cisco phone set to do NAT. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie alert: Cannot get voicemail to answer (have scoured the web for help)
Thanks for the help, but the problem remains:
did try the ,20 - same thing (but I tried 5 instead).
this was from the console:
-- Executing Wait("CAPI[contr1/444970]/5", "1") in new stack
-- started pbx on channel (callgroup=2)!
-- Executing Answer("CAPI[contr1/444970]/5", "") in new stack
-- CAPI Answering for MSN 444970
-- Executing DigitTimeout("CAPI[contr1/444970]/5", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("CAPI[contr1/444970]/5", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("CAPI[contr1/444970]/5", "tt-monkeysintro") in
new stack
-- Playing 'tt-monkeysintro' (language 'en')
== CDR updated on CAPI[contr1/444970]/5
-- Executing Dial("CAPI[contr1/444970]/5", "zap/1/711|5") in new stack
-- Called 1/711
-- Zap/1-1 answered CAPI[contr1/444970]/5
-- Hungup 'Zap/1-1'
== Spawn extension (default, 711, 1) exited non-zero on
'CAPI[contr1/444970]/5'
-- CAPI Hangingup
Julian.
> - Original Message -
> From: "Sean Cheesman" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, April 16, 2004 5:58 PM
> Subject: RE: [Asterisk-Users] Newbie alert: Cannot get voicemail to answer
> (have scoured the web for help)
>
>
> > > However, if there is no answer, or the extension is busy, *
> > > just keeps on trying to connect, and never drops to voicemail
> > > (busy or unavailable).
> > >
> > > exten => _7XX,1,Dial(zap/1/${EXTEN}|5m)
> > try something like exten => _7XX,1,Dial(zap/1/${EXTEN},20) where 20 is
> > the number of seconds you want it to time out after
> >
> > > exten => _7XX,2,Voicemail(u${EXTEN})
> > > exten => _7XX,3,Hangup
> > > exten => _7XX,103,Voicemail(b${EXTEN})
> > This should be the Dial priority plus 101 (or 102 in your case)
> >
> > Hope this helps
> >
> > Sean
> > ___
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>
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Re: [Asterisk-Users] Cisco 7940 no audio
It might be worth checking if there's a firewall in the way blocking your upstream audio and also the codecs setup on the cisco and in your asterisk box ?? There's not a lot of detail here to go on, sorry. On Fri, 16 Apr 2004, Craig Waddington wrote: > When we receive or make a call to the outside - they can hear us, but we > cant hear them. > > > > It may work 1 of 20 times. I have set canreinvite=no and looked at many > sites but cannot track down this problem. > > > > Current setup: > > > > Isdn Eicon Diva card / Capi -> Asterisk --> network. > > > > I have tried adjusting the RTP port in rtp.conf with the Cisco default > ports, no luck. > > > > Anyone had this problem, and has a fix? > > > > Thanks. > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 no audio
On Apr 16, 2004, at 10:04 AM, Craig Waddington wrote: When we receive or make a call to the outside – they can hear us, but we cant hear them. With SIP, missing audio is *usually* either a firewall or NAT issue. Check firewall logs and make sure that you aren't seeing packets being lost. Do you have more then one 7940? If so, can they call each other? Also, when people call into your system, do they get audio from asterisk? Does voicemail work? Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning from Asterisk
I have had problems with the NOTIFY packet being sent (not problems, just annoying warnings) when using keepalive with the Sipura SPA-2000. Asterisk will complain both using the Register and Notify methods (which are the two that the Sipura uses). The NOTIFY warning isn't actually causing any trouble, but I opted to just send no data (a carriage return will also work) to get the adapter to keep the port open. Sending blank info keeps asterisk from complaining as well. If you having this problem otherwise it is certainly something else. -Steve On Apr 16, 2004, at 1:20 PM, Kurt wrote: On the * console do a sip debug and look for the notify packet. It should give you a reason why it is being sent. Kurt __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning from Asterisk
On the * console do a sip debug and look for the notify packet. It should give you a reason why it is being sent. Kurt __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 no audio
When we receive or make a call to the outside – they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk à network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this problem, and has a fix? Thanks.
[Asterisk-Users] No read routine on channel AsyncGoto/Zap/1-1
I saw the error: No read routine on channel AsyncGoto/Zap/1-1 in my log today. Despite googling, I have no idea what this error relates to. Could someone please help me. Thanks JC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning from Asterisk
Hi there, Doesn't look serious to me. I get it when I use certain SIP phones. Can you make / receive calls ok ?? That's the bacon :) It probably means you have a phone that wants to tell asterisk about some functionality it has (or something). I think Olle is experimenting with code called chan_sip2.c I'm not sure if this includes notify support - think it might. You could do a search of the bug tracker. http://bugs.digium.com ?? I hope that helps. I'm sure someone else can give you more info on exactly what NOTIFY does, or you could check the RFC for SIP (as I'm still learning about SIP), but I can say if it's serious or not.. and my vote is that it shouldn't be / isn't. - Good luck! Chris On Fri, 16 Apr 2004, Joost Kraaijeveld wrote: > Hi all, > > I get this warning from Asterisk and I want to assess whether it is important, and > if so, if I should complain to the telephone manifacturer or start up my > programmer's editor: > > chan_sip.c:5152 handle_response: Host '172.31.1.7' does not implement 'NOTIFY' > > What does this mean and am I missing some important feature? > > Groeten, > > Joost Kraaijeveld > Askesis B.V. > Molukkenstraat 14 > 6524NB Nijmegen > tel: 024-3888063 / 06-51855277 > fax: 024-3608416 > e-mail: [EMAIL PROTECTED] > web: www.askesis.nl > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie alert: Cannot get voicemail to answer (have scoured the web for help)
> However, if there is no answer, or the extension is busy, *
> just keeps on trying to connect, and never drops to voicemail
> (busy or unavailable).
>
> exten => _7XX,1,Dial(zap/1/${EXTEN}|5m)
try something like exten => _7XX,1,Dial(zap/1/${EXTEN},20) where 20 is
the number of seconds you want it to time out after
> exten => _7XX,2,Voicemail(u${EXTEN})
> exten => _7XX,3,Hangup
> exten => _7XX,103,Voicemail(b${EXTEN})
This should be the Dial priority plus 101 (or 102 in your case)
Hope this helps
Sean
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Re: [Asterisk-Users] Matching variable-length extensions with chan_zap in overlap dialling
Try exten => _0X. <--- notice the period
Jeremy McNamara
Apollon Koutlides wrote:
I've been having trouble matching variable extensions on a zap channel
(an E1 line). Doing it the extensions.conf way:
[pri1]
; Match 8078078- calls
include => m807nat
include => m807mob
include => m807oth
[m807nat]
exten => _80780782X,1,StripMSD(7)
exten => _2X,1,SetVar,clidest=${EXTEN}
exten => _2X,2,Goto(cli,s,1)
[m807mob]
exten => _807807869,1,StripMSD(7)
exten => _69,1,SetVar,clidest=${EXTEN}
exten => _69,2,Goto(cli,s,1)
[m807oth]
exten => _80780780.,1,StripMSD(7)
exten => _0.,1,SetVar,clidest=${EXTEN}
exten => _0.,2,Goto(cli,s,1)
...when I dial, say, 00441565652244 * will match the first wildcard
digit immediately:
-- Accepting call from '2108126055' to '807807800' on channel 1, span 1
I've tried using an AGI to capture the rest of the digits, but that
didn't work either (wait for digit catches no digits), since the
channel is not answered yet (and I don't want to do that).
DigitTimeout in extensions.conf is of no consequence either, as long
as the call is not answered.
Looking in the bug archive I found this:
http://bugs.digium.com/bug_view_page.php?bug_id=0001422
which only perplexed me more... I tried to hack a bit of chan_zap (my
competence in C is far below adequate) and at least managed to avoid
matching immediately when there are more than one matches, but I got
stuck with the timeout issues :-)
Before trying (or rather paying somebody else with more programming
experience) to hack the chan_zap code to fit my needs, I thought I'd
consult the people on the list... any hints? Is there something I'm
missing here?
Apollon Koutlides
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RE: [Asterisk-Users] SoundPointR IP 300
I agree, the Polycom phones are real nice. I have one IP500 and will be purchasing about 10 more shortly. I tested the Snom as well as the GrandStream. The Snom is not a bad phone in general but the user interface is not as intuitive as the PolyCom, nor is the features as robust IMO. The grandstream did not work for me mainly because it lacked a 100Mbps switch so I would have to run separate 2 data jacks to every cube/office. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, April 16, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SoundPointR IP 300 It was my understanding that the SIP version is not available until May or June. My IP600's work great , though... -- Original Message -- From: Shad Mortazavi <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Fri, 16 Apr 2004 07:07:45 -0400 >Dear Group, > >Does any one have experience using SoundPoint(r) IP 300? > >I have one call center on Snom 200's I'm adding a second and was looking at >the SoundPoint, but needed some input. > >Thanks > >Shad Mortazavi >--- >Nexus Technical Manager >n|m Nexus Management Inc >Sydney > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning from Asterisk
Hi all, I get this warning from Asterisk and I want to assess whether it is important, and if so, if I should complain to the telephone manifacturer or start up my programmer's editor: chan_sip.c:5152 handle_response: Host '172.31.1.7' does not implement 'NOTIFY' What does this mean and am I missing some important feature? Groeten, Joost Kraaijeveld Askesis B.V. Molukkenstraat 14 6524NB Nijmegen tel: 024-3888063 / 06-51855277 fax: 024-3608416 e-mail: [EMAIL PROTECTED] web: www.askesis.nl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie alert: Cannot get voicemail to answer (have scoured the web for help)
I'm having a bit of a problem here:
I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a
x100p card for testing purposes.
As a proof of concept, I wanted to be able to dial into the * using the isdn
line, listen to a message, and enter a 3 digit extension number. If this
happens, I wanted the * box to dial out using the x100p card, into our PBX
(Nortel Meridian).
If there was no answer, I wanted to leave a voicemail message.
Now, I *can* dial in using isdn, enter the extension, and if someone
answers, I can speak to them with no problem.
However, if there is no answer, or the extension is busy, * just keeps on
trying to connect, and never drops to voicemail (busy or unavailable).
Please help - I have tried for hours now to find the problem. It is probably
a very obvious thing ...
I am using cvs head, downloaded this morning.
I have attached my conf files below.
Julian.
extensions.conf
[general]
static=yes
writeprotect=yes
[default]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,BackGround(tt-monkeysintro)
exten => s,6,BackGround(tt-monkeys)
exten => _7XX,1,Dial(zap/1/${EXTEN}|5m)
exten => _7XX,2,Voicemail(u${EXTEN})
exten => _7XX,3,Hangup
exten => _7XX,103,Voicemail(b${EXTEN})
exten => _7XX,104,Hangup
exten => #,1,Playback(demo-thanks)
exten => #,2,Hangup
exten => t,1,Goto(#,1)
exten => i,1,Playback(invalid)
exten => i,2,Hangup
capi.conf
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=OurNumberHere
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
;echosquelch=1
echocancel=yes
;echotail=64
callgroup=1
deflect=12345678
devices=2
zapata.conf
[channels]
language=en
signalling=fxs_ks
usecallerid=yes
echocancel=yes
channel => 1
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RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
> > Not knowing about the switch side of things, but how many interfaces > would a PRI card be able to handle in that switch? I'm betting for $35k > it is quite a few. It may be something to sit down with your copy of the > local tariffs and decide how many circuits over how many months would > pay off that card and make the business pitch to get it. > The card outputs 30 D channels. You still use 24 channels out of your IMT trunks, mapping channel 24 to one of the 30 D channels. Eventually I am sure they will get the card. They were hoping we could make things work in the short term some other way. Might just have to bite the bullet. > Your only other potential solution would probably involve SS7, and that > isn't supported under asterisk now. SS7 on asterisk would be nice for many reasons. Thanks for all your answers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 11:05, Mike Machado wrote: > My situation is a little different. The carrier switch is about 100 feet > away from my asterisk box. The company I am working for is a CLEC and > they have their own switch. The switch I am connected to does not have a > very expensive PRI signaling card ($35k), so they can only do CAS. > > If I was a customer and getting service from a carrier, I would > definitely have gotten a PRI. Thanks for your perspective on this. Not knowing about the switch side of things, but how many interfaces would a PRI card be able to handle in that switch? I'm betting for $35k it is quite a few. It may be something to sit down with your copy of the local tariffs and decide how many circuits over how many months would pay off that card and make the business pitch to get it. Your only other potential solution would probably involve SS7, and that isn't supported under asterisk now. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 08:45, mattf wrote: > Sadly it's a limited to PRIs(at least that's what I've been told) You just > can't send that much call data with a good old non-PRI T1. The up side is > you get one extra voice channel to use as compared to a PRI. It's strange > that the carrier doesn't have PRI cabability, I've run into many more > carriers that can't do non-PRI. I've even had one that couldn't send me > ANI(CallerID) on non-PRI lines. > > Once you get into the world of ISDN and PRI, you start to be able to do a > lot more with the signaling of calls, and you even have the ability to do > faster call switching as compared to a non-PRI. > > My advice is to get a different carrier, there are hundreds out there. And > if you absolutely have to have dynamic CallerID transmission you should > verify that your new carrier will let you do that before you sign a > contract(Make sure you verify it by talking to an actual switch tech from > the carrier, sales people will lie through their teeth to get you to sign > that contract). > > Hope this helps, > > MATT--- > My situation is a little different. The carrier switch is about 100 feet away from my asterisk box. The company I am working for is a CLEC and they have their own switch. The switch I am connected to does not have a very expensive PRI signaling card ($35k), so they can only do CAS. If I was a customer and getting service from a carrier, I would definitely have gotten a PRI. Thanks for your perspective on this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 10:46, Ryan Thrash wrote: > First, check zapata.conf to see what is in there. Next, I've not heard > of any luck with the name portion on T1s, but the number can be changed > for us. The name portion is no problem if you are on a PRI. I personally have had fun spoofing the callerid during some prank calls to my friends. In a channelized T1 like is mentioned below though, you don't have enough signaling to specify it. FATIK, you can oply specify the outgoing digits via some prearranged length and protocol similar to callerid and dnis on inbound calls. I'm not sure if asterisk supports this on channelized T1. > On Apr 16, 2004, at 10:30 AM, Mike Machado wrote: > > > > >>> > >> You can usually get CLI on an E&M robbed bit T1 by configuring it > >> right. > >> Instead of just sending you the DNIS as a string of DTMF they usually > >> send ***. The DNIS and CLI may be swapped, and there may be > >> less than 3 *s in the string - wonderful consistency, eh? :-\ > > > > I am getting CallerID and DNIS on the inbound calls. What I really need > > is to be able to set callerID on outbound calls. I am trying to set the > > callerid using SetCIDNum just before using Dial on a zap channel, but > > it > > looks like the switch guys have it set to always stamp the same > > callerID > > on the my outbound calls no matter what I put in SetCIDNum or what > > channel on the T1 I use. Is this a misconfiguration of the switch or a > > limitation of the signaling protocol? If its the switch, can you give > > me > > any pointers as to what I could ask them to look for, or if its the > > protocol, do you know any other signaling protocol that lets me set > > outbound callerID (besides PRI)? > -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
First, check zapata.conf to see what is in there. Next, I've not heard of any luck with the name portion on T1s, but the number can be changed for us. HTH, Ryan On Apr 16, 2004, at 10:30 AM, Mike Machado wrote: You can usually get CLI on an E&M robbed bit T1 by configuring it right. Instead of just sending you the DNIS as a string of DTMF they usually send ***. The DNIS and CLI may be swapped, and there may be less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
Sadly it's a limited to PRIs(at least that's what I've been told) You just can't send that much call data with a good old non-PRI T1. The up side is you get one extra voice channel to use as compared to a PRI. It's strange that the carrier doesn't have PRI cabability, I've run into many more carriers that can't do non-PRI. I've even had one that couldn't send me ANI(CallerID) on non-PRI lines. Once you get into the world of ISDN and PRI, you start to be able to do a lot more with the signaling of calls, and you even have the ability to do faster call switching as compared to a non-PRI. My advice is to get a different carrier, there are hundreds out there. And if you absolutely have to have dynamic CallerID transmission you should verify that your new carrier will let you do that before you sign a contract(Make sure you verify it by talking to an actual switch tech from the carrier, sales people will lie through their teeth to get you to sign that contract). Hope this helps, MATT--- -Original Message- From: Mike Machado [mailto:[EMAIL PROTECTED] Sent: Friday, April 16, 2004 11:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question > > > You can usually get CLI on an E&M robbed bit T1 by configuring it right. > Instead of just sending you the DNIS as a string of DTMF they usually > send ***. The DNIS and CLI may be swapped, and there may be > less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Windows Drivers for Wildcard FXO Card
And if you want to use it with windows telephony software, such as answering machine or modem communications software, you can probably take the drivers for the Intel MD3200 based modem, modify the .inf for the Digium vendor and device ID. I have not tried this, but since the MD3200 modem works that way in Linux, the X100P may work that way in Windows. Then you would have a $100 winmodem! Let us know what you find out. Jeremy -Original Message- From: Matteo Brancaleoni [mailto:[EMAIL PROTECTED] Sent: Friday, April 16, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Windows Drivers for Wildcard FXO Card /me set ban on *winzoz*drivers* they doesn't exist. wildcard are only for linux and only for asterisk. but you can port the driver to windows if you want Matteo. Il ven, 2004-04-16 alle 17:12, Bill McCready ha scritto: > Where may I find a Windows driver for a Wildcard FXO Card ??? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Windows Drivers for Wildcard FXO Card
In article <[EMAIL PROTECTED]>, Bill McCready <[EMAIL PROTECTED]> wrote: > Where may I find a Windows driver for a Wildcard FXO Card ??? Why would anyone want such a thing? -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
> > > You can usually get CLI on an E&M robbed bit T1 by configuring it right. > Instead of just sending you the DNIS as a string of DTMF they usually > send ***. The DNIS and CLI may be swapped, and there may be > less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Windows Drivers for Wildcard FXO Card
/me set ban on *winzoz*drivers* they doesn't exist. wildcard are only for linux and only for asterisk. but you can port the driver to windows if you want Matteo. Il ven, 2004-04-16 alle 17:12, Bill McCready ha scritto: > Where may I find a Windows driver for a Wildcard FXO Card ??? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Windows Drivers for Wildcard FXO Card
Where may I find a Windows driver for a Wildcard FXO Card ??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] errors on Pri
I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! J Any ideas. Apr 16 10:40:12 WARNING[213006]: chan_zap.c:5991 zt_pri_error: PRI: Read on 141 failed: Unknown error 500 Apr 16 10:40:12 NOTICE[213006]: chan_zap.c:6705 pri_dchannel: PRI got event: 6 on span 5 Apr 16 10:40:12 WARNING[213006]: chan_zap.c:5991 zt_pri_error: PRI: Read on 141 failed: Unknown error 500 Apr 16 10:40:12 NOTICE[213006]: chan_zap.c:6705 pri_dchannel: PRI got event: 6 on span 5 Apr 16 10:40:12 WARNING[213006]: chan_zap.c:5991 zt_pri_error: PRI: Read on 141 failed: Unknown error 500 Apr 16 10:40:12 NOTICE[213006]: chan_zap.c:6705 pri_dchannel: PRI got event: 6 on span 5 n Z What is event 6? Read on 141, is that a channel in zaptel.conf?? Oh and if an error is unknown why give it a number (500)??? I am running CVS from about a month ago. I updated the CVS that was running before but it did not solve the problem. This machine was running before with a T40O (4 port T1) and a T1000 (single port T-1) I pulled the T1000 and put it in another machine for a month or so to do some testing over IAX between machines. This span does talk to the PSTN via PRI. After I put it back in all the problems started. Could this be a case where the PCI PnP has ‘remembered’ the card and is causing a problem. All the usual stuff under /proc looks good and the cards are NOT sharing any interrupts. I get no other error messages that I can find. Ztcfg –vvv loads without errors and dmesg and all the other logs in /var/log look fine. If someone could tell me what these errors are that would be awesome and I would be willing to create and maintain the “What The F is this Error” page on the Wiki!
Re: [Asterisk-Users] PCPhoneline.com FXS to FXO Port Converter and SIPURA ATA
For users using the PCPhoneline.com FXS to FXO port converter with a Sipura ATA who experience DTMF detection problems from the PSTN, there is an input gain field located in the administration area of the Sipura ATA that may be adjusted to address this problem. The input gain field is often defaulted on the Sipura ATA to a negative value causing several decibels of volume loss such that the Sipura ATA cannot detect the incoming DTMF touchtones dialed remotely. To resolve the problem you should increase the value of the input gain field one step at a time and retest until the problem is resolved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Proble with sample.calla and setvar
Hi
I'm using the following sample.call
Channel: Zap/2
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: fromfax
Extension: 3
Priority: 1
SetVar: num=77
in extensions.conf I have:
[fromfax]
exten => 3,1,Wait,1
exten => 3,2,SayDigits(${num})
exten => 3,3,Hangup
But it only says 10 it seems variable is not set
do I use correctly SetVar ?
How can I read variables assigned using SetVar ?
Thank you
Bye
Lorenzo
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Re: [Asterisk-Users] FXO cards for TDM400P....
On Thu, Apr 15, 2004 at 11:14:37PM -0500, Steven Sokol wrote: > > Is there any word on the availability of the FXO cards for the TDM400P? > > I have an application that would benefit. If it has been dropped please > > let me know. > > Word has it that they should hit distributors in the next week or perhaps > two. One caveat -- they do not have FCC certifications yet. I have a pair > of them backordered from my distributor. So don't give up. Just a few more > days. Along the same lines, what about the 16-port FXS/FXO card that was alluded to about 2 months ago? IIRC, it was to be released in about 6 weeks (or about 2 weeks ago now...). I'm not as interested as I was, as I've gone the channel bank route now, but I'm still very interested in seeing more information about it... Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. Jesus saves, Allah forgives, Cthulhu thinks you'd make a nice sandwich. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application
interested ... please send me some info. - Original Message - From: "Kyle Hagan" <[EMAIL PROTECTED]> Date: Thu, 15 Apr 2004 09:20:01 -0700 To: <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] PC based Switchboard application > Im interested can you send information? > > Kyle > [EMAIL PROTECTED] > > > - Original Message - > From: "Pertti Pikkarainen" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, April 10, 2004 2:26 AM > Subject: Re: [Asterisk-Users] PC based Switchboard application > > > > We have switchboard application ( PC+browser+Java ) with quite a rich > > feature set. > > It talks to * via manager port. > > Works as a call center too. > > However, it is not open source. > > If you are interested in, please contact me directly. > > > > Best regards Pertti > > > > Keith D'Atrio wrote: > > > > > Hello All > > > I am looking for a PC based switchboard application. Cisco > > > CallManager has a web attendant console that allows you to use the PC > > > to transfer calls and the like and I was wondering if there was a > > > similar program compatible with *. > > > Thank you in advance > > > Keith D'Atrio > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] t1 won't dial outbound
Joe,
Thanks for your response...if nothing else this is definitely going to
help simplify my layout. I'm redoing my zapata file now and going to
try it out. Thanks again.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, April 15, 2004 2:52 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] t1 won't dial outbound
I haven't tried breaking up the channels into different groups (mainly
because I haven't had a need to), but the examples I've seen looked more
like:
[channels]
signalling=em_w
switchtype=5ess
group=1
context=uti-mainst
channel => 1-3
group=2
context=sales
channel => 4-6
group=3
etc...
In this example, the signalling and switchtype don't change (because
they are all on the same trunk), but you can change the context in each
group definition. Anything specified ABOVE the channel statement will
be applied to those channels. So, you only need to specify the changes
inbetween your channel => statements.
As such, all of the other statements before the channel => 6 statement
will also be applied to that channel. If you specified a parameter
(like callretrun=yes or callprogress=yes) that the LEC (Carrier) didn't
like, it would not accept the call. If group 5 works correctly for
outbound calls, I would model group 3's defninitions after group 5.
Joe
"Mark Messmore, Technical Support, University Telcom Inc."
<[EMAIL PROTECTED]> wrote:
Thanks for the reply.
I didn't include my entire zapata.conf...just the portion that applied
to this call (i.e. group #3)
Please correct me if I have misunderstood how this all works together.
When I see:
-- Executing Dial("SIP/mark-2d08", "Zap/g3/2550559") in new stack
-- Called g3/2550559
-- Hungup 'Zap/6-1'
I'm interpreting that this is dialing out on Zap group 3 (which happens
to begin on channel 6). Please correct me if I'm wrong here...
I'm attaching my entire zapata.conf just to defer any confusion...and
to see if you can see anything.
Also, I'm going to take your suggestion and create another zapata.conf
which will be simplified just to see if there is a conflict somewhere in
there.
Thanks for your help!
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, April 15, 2004 1:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] t1 won't dial outbound
It looks like your channel and group statements in the zapata.conf are
the problem. Notice that when it tries to dial out it does so on
Zap/6-1. You have the T-1 defined as 'Span 1,' but you are trying to
send the calls to span 6. It ain't gonna work! I don't see anywhere
where you've assigned the rest of the channels on that T-1, either. I
would recommend either grouping them all together (that's the easiest),
or at least making sure you've got all of the channels assigned to
groups. My zapata.conf is much simpler:
signalling=pri_net
group=1
channel => 1-23
When it dials, then you will see the calls going out on Zap/1-1 or
Zap/1-2, etc.
Good luck; and have fun!
Joe
"Mark Messmore, Technical Support, University Telcom Inc."
<[EMAIL PROTECTED]> wrote:
I've posted this problem a couple of times before with little or no
response. Basically I have a T100P in my * box. Incoming calls are
working great. However outgoing calls are not working at all. I've
copied a previous post into this message which should have all the
necessary info. Any ideas or suggestions would be greatly appreciated.
Thanks.
Mark
#
OK...I've got an * box with a T100P in it. For the most part incoming
calls are going through just fine. Outgoing calls, however, I'm having
some more trouble with. Whenever I make an outgoing call, the call
begins, however after the dialing process all I hear is dead air.
Here's the output from my * console:
-- Executing Dial("SIP/mark-2d08", "Zap/g3/2550559") in new stack
-- Called g3/2550559
-- Hungup 'Zap/6-1'
== Spawn extension (uti-mainst, 2550559, 1) exited non-zero on
'SIP/mark-2d08'
I've checked with the switch guy...and whatever channel I'm trying to
dial out on is coming up as "blocked" on his switch. We've compared as
many settings as we can think of and they all seem to be set the same.
I'll post the entries from my zaptel.conf and my zapata.conf in
here...if you have any ideas please send them my way...
zaptel.conf
span=1,1,0,d4,ami
e&m=1-24
fxsks=25
loadzone=us
defaultzone=us
zapata.conf
context=conference
signalling=em
switchtype=5ess
group=3
callgroup=3
pickupgroup=3
channel => 6
busydetect=yes
callerid=asreceived
callprogress=yes
callreturn=yes
callwaiting=yes
callwaitingcallerid=yes
cancallforward=y
Re: [Asterisk-Users] SoundPointR IP 300
It was my understanding that the SIP version is not available until May or June. My IP600's work great , though... -- Original Message -- From: Shad Mortazavi <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Fri, 16 Apr 2004 07:07:45 -0400 >Dear Group, > >Does any one have experience using SoundPoint(r) IP 300? > >I have one call center on Snom 200's I'm adding a second and was looking at >the SoundPoint, but needed some input. > >Thanks > >Shad Mortazavi >--- >Nexus Technical Manager >n|m Nexus Management Inc >Sydney > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice lenght option
I would like to know if there is an option that can limit the amount of total minutes that an individual mail box can hold. Kurt __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem
Hello, From: "Joe Dennick" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Strange T1 Problem Date: Fri, 16 Apr 2004 07:44:17 -0500 Can one use a pipe '|' for the Dial application the same way that one would use a comma ','? I know this one works, but what I don't know is if it will also work using pipes in place of the commas. Joe Yes, You can use '|' for the dial application. In fact even if you use comma(,) in your extensions.conf, Asterisk replaces it with '|' when it builds the dial plan. See the following entry in extensions.conf: [test] exten => 1234,1,Dial(SIP/1234,20,r) exten => 1234,2,Voicemail(u1234) and the dialplan for this is: * CLI> show dialplan test [ Context 'test' created by 'pbx_config' ] '1234' => 1. Dial(SIP/1234|20|r) [pbx_config] 2. Voicemail(u1234) [pbx_config] Regards, Girish _ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy problems (was music on hold problems)
I definitely have the correct usb card. i have another machine running fedora core 1 (kernel 2.4.22) and everything works no problem. i remember reading somewhere that there were some driver updates in 22 that resolved some problems. seems like I could be running into that. -steve On Apr 15, 2004, at 6:52 PM, Iain Stevenson wrote: --On Thursday, April 15, 2004 6:43 pm -0400 Steven Kokinos <[EMAIL PROTECTED]> wrote: Actually, after rebooting my machine music on hold started working properly. Not sure what the issue was. As for ztdummy, I am having a more substantive issue with that, which is keeping me from getting meetme working. however, usb-uhci.o does in-fact exist. Does anyone have any thoughts? You have an appropriate USB card installed? - ztdummy won't work with ohci cards. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX firmware for snom 200s?
is there a firmware for IAX for the snom 200's. or are there any other hard phones that use iax(2)? Thanks in advance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: sorry ... list problem
Could the moderator please check my e-mail address. I stopped getting messages from this list. I did send an e-mail to [EMAIL PROTECTED] and it got bounced back from the remote mail server. - A.G. Edwards & Sons' outgoing and incoming e-mails are electronically archived and subject to review and/or disclosure to someone other than the recipient. -
[Asterisk-Users] SIP IAX2 MySQL Config
I've configured asterisk to connect a MySQL database for CDR, Voicemail and SIP/IAX2 peers. - CDR are reccorded - Voicemail config is readen directly in the database but SIP/IAX2 peers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 192.168.0.10:5060 == Using TOS bits 0 Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user' == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr 16 14:10:34 WARNING[1074449120]: chan_iax2.c:6218 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 192.168.0.10 port 4569 What is the problem? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropped calls
> Ok, I'll watch for that as well since I upgraded my desk's Grandstream to > 1.0.4.54 an hour ago (previously I had either 4.26 or 4.17). It appears the hangup is triggered by a SIP ACK with CSeq set to 0. Some Grandstream UAs happen to pick 0 as CSeq. chan_sip.c contains if (!p->lastinvite && !strlen(p->randdata)) p->needdestroy = 1; where p->lastinvite hold the matching CSeq from the last INVITE. 0 in this case... Fixing this does bring downs the number of hangups, but does not entirely solve the problem. We are still looking. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange T1 Problem
Can one use a pipe '|' for the Dial application the same way that one
would use a comma ','?
My dialplan looks like this:
exten => 1234,1,Dial(SIP/1234,20,r)
exten => 1234,2,Voicemail(u1234)
etc
I know this one works, but what I don't know is if it will also work
using pipes in place of the commas.
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, April 16, 2004 7:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Strange T1 Problem
Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well. It will just ring
forever.
Cheers,
Willy
- Original Message Follows -
>
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
>
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently.
>
> This must be a different problem, because neither of those solutions
> worked.
>
>
>
> zapata.conf sends call to fixup context:
>
>
> [fixup]
>
> ; Receive call as **
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
>
>
> [default]
>
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
>
>
> user1's phone rings, but no ring from PSTN caller. user1 picks up,
> both can talk ok.
>
>
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
>
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ypOne Publishing
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Re: [Asterisk-Users] TE405P + Adit 600 and FXO module - should this work?
Thanks! I'll spend some time with this information, check the pinouts and I should be well on my way. Glad to know it can work! ;-) -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: "Darren Nickerson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, April 16, 2004 1:29 AM Subject: [Asterisk-Users] TE405P + Adit 600 and FXO module - should this work? > Folks, > > I'm experimenting with bringing multiple (8) analog lines from our local > telco into a Carrier Access ADIT 600 channel bank with an FXO module, then > having this talk to Asterisk via the T1 TDM controller on the ADIT and a > TE405P card. > > I don't know if this will work well (ie: give me decent echo cancellation, > call disconnect supervision, caller ID etc) but I haven't had much luck > getting this combination flying at all thus far. Should it work? > > I've been concentrating on the Adit T1 interface <> TE405P connection > ... and have the following in zapata.conf: > >span=1,1,0,esf,b8zs >fxsks=1-24 >loadzone = us >defaultzone=us > > At the end of my zaptel.conf, I have: > >signalling=fxs_ks >group=2 >callerid="Joe Schmoe" <(215) 555-1212> >channel => 1-24 > > To get things flying, I do: > >modprobe zaptel >modprobe wct4xxp > > which causes the TE405P card to activate, but show a single flashing red > alarm on the configured span. > > The Adit's TDM controller also displays a solid red LED. Here's its status: > >Adit 600> status a:1 >SLOT A: >Status for DS1 1: >Receive: Loss of Signal >Transmit:RAI/Yellow Alarm >Loopback:OFF > >Adit 600> > > I'm not sure I have the DS1 configured appropriately. It says: > >Adit 600> show a:1 >SLOT A: >Settings for DS1 1: >Circuit ID: CAC DS1# A:1 >Up/Down: UP >Framing: ESF >Line Coding: B8ZS >Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB) >Loop Code Detection: ON >Loopback:OFF >FDL Type:None > > Can anyone familiar with the Adit 600 and/or TE405P see any obvious errors > here? > > -Darren > > -- > Darren Nickerson > Senior Sales & Support Engineer > iFAX Solutions, Inc. www.ifax.com > [EMAIL PROTECTED] > +1.215.438.4638 ext 8106 office > +1.215.243.8335 fax > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem
Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well. It will just ring
forever.
Cheers,
Willy
- Original Message Follows -
>
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
>
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently.
>
> This must be a different problem, because neither of those
> solutions worked.
>
>
>
> zapata.conf sends call to fixup context:
>
>
> [fixup]
>
> ; Receive call as **
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
>
>
> [default]
>
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
>
>
> user1's phone rings, but no ring from PSTN caller. user1
> picks up, both can talk ok.
>
>
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
Willy Wouters
ypOne Publishing
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Re: [Asterisk-Users] TE405P + Adit 600 and FXO module - should this work?
I have an Adit 600 working with a TDM400p card. - check this link for info on Adit-600 configuration http://lists.digium.com/pipermail/asterisk-users/2003-June/013072.html -Make sure the pinouts for your T1 cable are correct: DigiumT1 -> RJ-48(F) X-over Digium T1 Card Channel Bank RJ-48(F) Pin Pair#T/R Color Pin T/R 12 T white/orange 4 R1 22 R orange/white 5T 41 T blue/white 1 R 51 R white/blue 2 T1 3, 6, 7, 8 Not assigned Darren Nickerson wrote: Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much luck getting this combination flying at all thus far. Should it work? I've been concentrating on the Adit T1 interface <> TE405P connection ... and have the following in zapata.conf: span=1,1,0,esf,b8zs fxsks=1-24 loadzone = us defaultzone=us At the end of my zaptel.conf, I have: signalling=fxs_ks group=2 callerid="Joe Schmoe" <(215) 555-1212> channel => 1-24 To get things flying, I do: modprobe zaptel modprobe wct4xxp which causes the TE405P card to activate, but show a single flashing red alarm on the configured span. The Adit's TDM controller also displays a solid red LED. Here's its status: Adit 600> status a:1 SLOT A: Status for DS1 1: Receive: Loss of Signal Transmit:RAI/Yellow Alarm Loopback:OFF Adit 600> I'm not sure I have the DS1 configured appropriately. It says: Adit 600> show a:1 SLOT A: Settings for DS1 1: Circuit ID: CAC DS1# A:1 Up/Down: UP Framing: ESF Line Coding: B8ZS Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB) Loop Code Detection: ON Loopback:OFF FDL Type:None Can anyone familiar with the Adit 600 and/or TE405P see any obvious errors here? -Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupting Dial / Qwest-like transfers
> 2) While I'm at it...has anyone called Qwest lately? When they transfer > you, they can stay on the line the entire time, listen with you while it > rings, then when it says "Press 1 for XXX", then they can dial it for you, > talk to whoever answers and say "I've got Ryan on the line and needs > [whatever]", then leave the conversation and let you talk to the new > person. This whole time you're able to sit and listen to what's going on or > chat with the first person that's doing all this work for you. Is this > possible with Asterisk?! It's called assisted transfer and no, asterisk currently cannot do that. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SoundPointR IP 300
Title: SoundPointR IP 300 Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney
[Asterisk-Users] Matching variable-length extensions with chan_zap in overlap dialling
I've been having trouble matching variable extensions on a zap channel
(an E1 line). Doing it the extensions.conf way:
[pri1]
; Match 8078078- calls
include => m807nat
include => m807mob
include => m807oth
[m807nat]
exten => _80780782X,1,StripMSD(7)
exten => _2X,1,SetVar,clidest=${EXTEN}
exten => _2X,2,Goto(cli,s,1)
[m807mob]
exten => _807807869,1,StripMSD(7)
exten => _69,1,SetVar,clidest=${EXTEN}
exten => _69,2,Goto(cli,s,1)
[m807oth]
exten => _80780780.,1,StripMSD(7)
exten => _0.,1,SetVar,clidest=${EXTEN}
exten => _0.,2,Goto(cli,s,1)
...when I dial, say, 00441565652244 * will match the first wildcard
digit immediately:
-- Accepting call from '2108126055' to '807807800' on channel 1, span 1
I've tried using an AGI to capture the rest of the digits, but that
didn't work either (wait for digit catches no digits), since the channel
is not answered yet (and I don't want to do that).
DigitTimeout in extensions.conf is of no consequence either, as long as
the call is not answered.
Looking in the bug archive I found this:
http://bugs.digium.com/bug_view_page.php?bug_id=0001422
which only perplexed me more... I tried to hack a bit of chan_zap (my
competence in C is far below adequate) and at least managed to avoid
matching immediately when there are more than one matches, but I got
stuck with the timeout issues :-)
Before trying (or rather paying somebody else with more programming
experience) to hack the chan_zap code to fit my needs, I thought I'd
consult the people on the list... any hints? Is there something I'm
missing here?
Apollon Koutlides
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[Asterisk-Users] * <-> FWD behind NAT
Hello all, My * box is up and running but the only problem is I can't send or receive calls via my FWD account. I have tried everything I can think of to fix this, but no luck. Does anyone have any tips for my sip.conf. I am running * behind a nat'd router/firewall. Thanks in advance, Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: Re: [Asterisk-Users] Re: External access to voicemail
I am interested too so I think it would a good thing to post an URL (as you said) Thank you![EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«±:5%H$HJ+ºZµê)¶*'²ø¬Øm¶ÿ+-±Ø é¢oæj)fjåËbú?jË^®+$ºÇ«
