[Asterisk-Users] Upgrade to [EMAIL PROTECTED] 1.3 Problems with Background files
Title: Upgrade to [EMAIL PROTECTED] 1.3 Problems with Background files Hi All, I have upgraded from asterisk 0.7 to [EMAIL PROTECTED] 1.3 and found the following problems: I use distinctive ring to direct calls to the office or home. I have 3 gsm file for 3 scenarios, this now does not work. Office hours gsm After hours gsm Home gsm Symptoms If I have 1 gsm file in the exenstion.conf all works fine, no matter which file. But when I have 2 or more files I will not hear the voice of the second file added, only silence. But all the standard sound work?? Has anyone had this problem. I have reinstalled [EMAIL PROTECTED] incase it was something simple, with no luck. Regards Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mental Blank: HELP: I cant get any callerid on capi incoming?? WHY
All, Capi config ok, but I cant get a incoming number displayed on the screen of the snoms; only asterisk appears. What do I need in my extension.conf to make it display the number received?? Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AstLinux - New Version - w/ 1.0.3 what about capi!!!!
That's great, But does any one know of a package that has capi as part of it. ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, 16 December 2004 3:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AstLinux - New Version - w/ 1.0.3 Hello everyone, I have posted a new version of AstLinux (0.1.6). This one should be the best yet (relatively speaking). I have made a lot of init changes, added some applications, and of course - updated to asterisk-1.0.3, zaptel-1.0.3, and libpri-1.0.3. You may also like the improved website (don't expect much more) ;). Find more below: http://www.krisk.org/astlinux/ On another note, I need to hear back from people with success stories, problems, etc. As of now, there have been almost 1000 downloads of the various versions of AstLinux, but I have had only a couple of people contact me with questions or problems. I know that it isn't perfect, and that is why I need the people out there to help me find problems so that I can fix them (hopefully). Don't be shy! This is your chance to complain! (Just try not to e-mail me all at once) :) -- Kristian Kielhofner ___ 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] Caller ID on Snom 190?
In the phones web browser is an area that describes caller id as name or number display, Regards Michael Hatzis 0421 476 211 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, 14 December 2004 5:34 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Caller ID on Snom 190? Has anyone had success with the Snom 190 displaying caller ID name and number on the Snom 190 on for an inbound call from *? Right now our Snom's only show the caller id name, not number. I know the number is transmitted from the Telco and received by * since the number shows on the incoming call event at the * console. We are not setting the caller id in the extensions.conf, simply passing on what * receives from the PRI (via a single span Digium board). Damon ___ 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] Busy message on ISDN cards?
I had the same problem even though it was with capi, this may help. Have
you set your msn as Andrew or your line number??
Try this
exten = 2468,1,Dial(${TRUNK}/91234567:0412345678:1)
Regards
Michael Hatzis
0421 476 211
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Furey
Sent: Tuesday, 14 December 2004 4:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Busy message on ISDN cards?
Hi all,
I'm new to asterisk and not too knowledgeable on ISDN, so please be
gentle :)
I have a dual-channel Eicon Diehl Diva card in a Debian Woody box with
kernel 2.4.27, connecting to a Telstra (Australia) Onramp Home Highway
ISDN line. I'm pretty certain the card and line both work since
they've been used in this machine for PPP before this (but with an
older kernel with DoV patches, which are no longer to be used).
If I do
# modprobe hisax type=11,11 protocol=2,2 id=HiSax
it responds (in the syslog) with:
kernel: ISDN subsystem Rev:
1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded
kernel: HiSax: Linux Driver for passive ISDN cards
kernel: HiSax: Version 3.5 (module)
kernel: HiSax: Layer1 Revision 1.1.4.1
kernel: HiSax: Layer2 Revision 1.1.4.1
kernel: HiSax: TeiMgr Revision 1.1.4.1
kernel: HiSax: Layer3 Revision 1.1.4.1
kernel: HiSax: LinkLayer Revision 1.1.4.1
kernel: HiSax: Total 2 cards defined
kernel: HiSax: Card 1 Protocol EDSS1 Id=HiSax (0)
kernel: HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2
kernel: PCI: Found IRQ 9 for device 00:09.0
kernel: PCI: Sharing IRQ 9 with 00:04.2
kernel: Diva: IPAC PCI card configured at 0xd0862000 IRQ 9
kernel: Diva: IPAC PCI space at 0xd086
kernel: Diva: IPAC version 1
kernel: Eicon.Diehl Diva: IRQ 9 count 1697
kernel: Eicon.Diehl Diva: IRQ 9 count 1705
kernel: HiSax: DSS1 Rev. 1.1.4.1
kernel: HiSax: 2 channels added
kernel: HiSax: MAX_WAITING_CALLS added
so it appears to be detected. I'm using the following modem.conf:
[interfaces]
context=remote
driver=i4l
language=en
type=autodetect
dialtype=tone
mode=immediate
group=1
msn=91234567
incomingmsn=*
device = /dev/ttyI0
Starting asterisk with -c returns:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_modem.so] = (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulated Modem
Driver)
But if I define a test extension such as:
TRUNK=Modem/g1
exten = 2468,1,Dial(${TRUNK}/91234567:0412345678)
and try to dial it, the console says:
Dec 14 13:29:17 WARNING[15375]: chan_modem_i4l.c:608 i4l_dial:
Outgoing MSN andrew not allowed (see outgoingmsn=,, in modem.conf)
-- Called g1/91234567:0412345678
-- Modem[i4l]/ttyI0 is busy
-- Hungup 'Modem[i4l]/ttyI0'
I gather than busy is used for pretty much everything except for no
connection, but are there any suggestions of where to look?
Thanks in advance,
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
-- Bill Garrett
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
Hi, 12.2. most have bugs. You need to check version. Also you may want to try setting up a second voice codec and add alaw / ulaw as your first preferences. This may work? But I think your biggest problem is your ios version. Ps Don't forget to add you new voice codec preferences under your voice peer. Regards Michael Hatzis 0421 476 211 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Monday, 13 December 2004 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi thanks for your help . I do not have direct access to the Cisco, but I believe that he is AS5300 The ios version is 12.2 and the cisco dum config is: GWSCZ01en Password: GWSCZ01#sh run Building configuration... Current configuration : 5053 bytes ! ! Last configuration change at 05:17:58 UTC Mon Apr 16 2001 ! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001 ! version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname GWSCZ01 ! no boot startup-test logging queue-limit 100 ! ! ! resource-pool disable spe default-firmware spe-firmware-1 ip subnet-zero ip cef no ip domain lookup ! isdn switch-type primary-net5 ! ! voice service voip fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none sip ! voice class codec 11 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 gsmfr codec preference 4 g726r32 codec preference 6 g726r16 codec preference 7 g723r63 codec preference 8 g723r53 codec preference 9 g726r24 codec preference 10 g723ar63 codec preference 11 g723ar53 codec preference 12 g711ulaw codec preference 13 g711alaw codec preference 14 clear-channel ! ! ! ! ! ! ! no voice hpi capture buffer no voice hpi capture destination ! voice source-group cisco access-list 8 carrier-id target cisco ! ! ! fax interface-type fax-mail mta receive maximum-recipients 0 ! ! ! controller E1 7/0 framing NO-CRC4 line-termination 75-ohm ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled cas-custom 0 country bolivia ! controller E1 7/1 line-termination 75-ohm pri-group timeslots 1-31 ! controller E1 7/2 line-termination 75-ohm pri-group timeslots 1-31 description Embratel --More-- ! ! interface FastEthernet0/0 ip address y.y.y.y 255.255.255.224 duplex auto speed auto no cdp enable h323-gateway voip interface h323-gateway voip id GK01 ipaddr y.y.y.z 1719 h323-gateway voip h323-id GWSCZ01 h323-gateway voip tech-prefix 2032# ! ! ip classless ip route 0.0.0.0 0.0.0.0 y.y.y.v no ip http server ! ! ! ! ! ! call rsvp-sync ! voice-port 7/0:0 compand-type a-law ! voice-port 7/1:D ! voice-port 7/2:D ! voice-port 7/3:0 compand-type a-law ! voice-port 7/4:0 compand-type a-law ! voice-port 7/5:0 ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice voip destination-pattern 44T voice-class codec 11 session protocol sipv2 session target sip-server session transport udp ! dial-peer voice pots destination-pattern T direct-inward-dial port 7/0:0 ! sip-ua retry invite 3 retry cancel 2 sip-server ipv4:x.x.x.x ! On Sun, 2004-12-12 at 20:07, Hatzis, Michael wrote: What's the cisco box,52 / 53; version ios? can you post a config dump? Regards Michael Hatzis 0421 476 211 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Monday, 13 December 2004 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Excuse the insistence but I am more than one week with this problem, and I do not have any idea to solve it. You know if the configuration with GK in the Cisco, can be interfering with the RTP traffic? Thanks in advance On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote: Pls, post your Cisco and * config files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Friday, December 10, 2004 12:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A) Cisco AS5xxx ASteriskPSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains 19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974
[Asterisk-Users] BRI Problem dialing out
Hi All, I have a slight problem when trying to dial out. When I dial any number out I get only a dial tone and the number is not dialed I have to then dial it manually. I have tried my extension.conf with my pstn box and it works fine but for some reason it wont with the isdn card. Im using the fritz pci card. Has any one else had this problem in the past??. I have also tried to set up an extension that will open the line then SendDTMF of a number to dial. no luck Can any one help Regards Michael Hatzis ___ 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 AS5XXX to asterisk debugging.
What's the cisco box,52 / 53; version ios? can you post a config dump? Regards Michael Hatzis 0421 476 211 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Monday, 13 December 2004 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Excuse the insistence but I am more than one week with this problem, and I do not have any idea to solve it. You know if the configuration with GK in the Cisco, can be interfering with the RTP traffic? Thanks in advance On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote: Pls, post your Cisco and * config files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Friday, December 10, 2004 12:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A) Cisco AS5xxx ASteriskPSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains 19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.740415 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.810312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.860314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.980351 IP (tos 0x0, ttl 64, id 180, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.000313 IP (tos 0x0, ttl 64, id 181, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none], proto 17, length: 164) y.y.y.y.18975 x.x.x.x.19927: UDP, length 136 19:06:01.020312 IP (tos 0x0, ttl 64, id 182, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.040302 IP (tos 0x0, ttl 64, id 183, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.060343 IP (tos 0x0, ttl 64, id 184, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.083311 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.128314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.130316 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.165318 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.186312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 Where x.x.x.x = ip address of Astersik y.y.y.y = ip address of Cisco Two types of codecs were proven ( ulow, g729 ). When use the Asterisk with Sip phones everything works well. SipPhone--Asterisk---PSTN(B) The configurations, are the usual ones (from the wiki). the version of asterisk is 1.0.3, the linux is FC2. Please help me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
