RE: [Asterisk-Users] Re: Upgrade from Altigen
If you're talking plain FXO/FXS ports (as in two-wire CO lines, not T1 or anything fancy), go LOW tech: Pick up a DPDT relay with 12V or 5V coil-voltage from your favorite electronics outlet (allelectronics, jameco, etc), and hook it up like so: Phone | / \ FXS FXO/CO When the coil is energized, the phone will be connected to the FXS port; when power is absent, the phone will be connected to the CO-line (which is still connected to the FXO port on your PBX. Now connect the coil to the appropriate power on your PBX computer... When the computer is on, the phone is an extension on the PBX; when the computer is off, your phone is connected directly to the CO line for emergencies. I have this working on my home-PBX w/o a problem; the relay cost me $2 or $3 incl. shipping (with some other parts), the connectors were strewn about my office. -Original Message- From: Geoff Nordli [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 27, 2004 10:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Upgrade from Altigen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sent: Tuesday, July 27, 2004 7:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Upgrade from Altigen Thanks Jim. Does anyone think that the Altigen has this feature built-in or might they have a device similar to the Power Fail Bypass installed? The Altigen web site has technical manuals online. http://www.altigen.com/customer_tech-manuals.html It would appear that the power failure transfer feature is part of their hardware. I believe that in some places there are legal requirements that some ability to call emergency services remain during a power failure. Jim James H. Thompson [EMAIL PROTECTED] Thanks Jim. I found the info in the guide. Apparently upon failure the card will automatically switch an incoming trunk to the first extension on the card. So this brings me to find a similar solution. I noticed that VoiceTronix OpenSwitch has this to say on their site: Phones Function on Power or PC Failure * Loop-Start ports switch through to Station ports on power or PC failure to preserve basic telephone functionality. * This by-pass mechanism connects the external PSTN lines to telephone handsets on station ports. * A watch-dog timer triggers the by-pass mechanism on PC failure. From how I read this the card will provide failover the to the FXS devices. There is no documentation on the web site. Can some expand on how this works? I assume that I can manually configure FXO to FXS mapping. It doesn't look like the Digium hardware supports this feature. Is that right? Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] drivers, kernel 2.6 and distribution
On Mon, 2004-07-26 at 14:33, Leif Madsen wrote:
On Mon, 26 Jul 2004 13:58:04 -0700, Florin Andrei
[EMAIL PROTECTED] wrote:
On Mon, 2004-07-26 at 13:12, Leif Madsen wrote:
ztdummy works fine on FC2. I was able to get a TDM400P to work first
try.
Using the distro kernel, or the vanilla 2.6?
Distro kernel.
Alright. For the record, in case others are following the same path,
here is what i did:
So this is a Fedora 2 system, fully updated, running kernel
2.6.6-1.435.2.3. The machine is a single-CPU AthlonXP.
Install the kernel-sourcecode package, make the symlink:
lrwxrwxrwx 1 root root 30 Jul 27 20:23 /usr/src/linux-2.6 -
/usr/src/linux-2.6.6-1.435.2.3
Go to /usr/src/linux-2.6, edit Makefile and change EXTRAVERSION from
-1.435.2.3custom to -1.435.2.3
Save Makefile. Run make menuconfig, change nothing, exit saving the
config. Run make and wait for kernel components to compile.
This ends the preparation stage.
Download the Asterisk 1.0-RC1 RPMs from here (the Fedora 1 packages
since there are no Fedora 2 packages there yet):
ftp://ftp.nacs.net/asterisk
Unpack the zaptel src.rpm (rpm -ivh zaptel...src.rpm), go to
/usr/src/redhat/SPECS, edit zaptel.spec so that the make is changed
into a make linux26 (i could probably automate that, so the package
builds correctly regardless of the kernel, but that's not my goal):
%build
make KINCLUDES=/lib/modules/%{kversion}/build/include
KSMP=%{?ksmp:-D__SMP__} \
ECHO_CANCELLER=-DECHO_CAN_MARK2 linux26
^^^
Save the spec, then build the package:
rpmbuild -ba zaptel.spec
Install the zaptel and kernel-module-zaptel packages. Run depmod -a
just in case.
Run modprobe zaptel. Run modprobe wcfxs. Both commands yield no
errors whatsoever on my system.
lsmod displays:
Module Size Used by
wcfxs 32032 0
zaptel219012 1 wcfxs
dmesg displays:
Zapata Telephony Interface Registered on major 196
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
All LEDs on the Wildcard are lighted green. Since my server also has two
dual-port Intel Pro/100 NICs (total 4 Ether ports), now the back of the
system looks like a Borg cube control panel. :-)
So far so good. I didn't run any hardware tests yet, but the results so
far are encouraging.
Thanks for the hints.
--
Florin Andrei
http://florin.myip.org/
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[Asterisk-Users] E-mail address for DIAX support has been changed
Hi all, Because of the huge amount of spam, I have been forced to change my e-mail address used for DIAX support. From now on, please use danto-at-rdslink-dot-ro (the one from this e-mail) instead of [EMAIL PROTECTED] In the same time, in order to get some more functionality for my cable link, the IP address of my Asterisk box has been changed by the ISP, so the CallMe functionality in DIAX is no more available till the next version of DIAX will be released. Thank you for your understanding and best regards. Dan ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
I have a Sipura-3000 and am hoping to use it to provide FXS/FXO ports for my Asterisk box. I don't have it working well yet but I blame that on my inexperience with Asterisk. Some configuration examples are available at http://voxilla.com/forum-viewtopic-t-557-sid-97ab81ff1df626865dd84ab79b4cd7d8.html or http://tinyurl.com/5nwum. -- Greg Broiles, JD, EA [EMAIL PROTECTED] (Lists only. Not for confidential communications.) Law Office of Gregory A. Broiles San Jose, CA ___ 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: OT: Re: [Asterisk-Users] John Vogel
Also, if you end up publishing it yourself, LaTeX will provide you with great features which will make your life MUCH easier and your work MUCH more professional-looking. Personally I write all my documents in LaTeX using vim but that is strictly a matter of personal preference. I wrote my manuals also in LaTeX, but not with vi or vim, but with a Wiki, CGI::KWiki. I wrote a python script kwiki2latex that does the conversion for me. It can handle pictures, automatic TOC, half-automatic Index, Tables, chapter-to-subsubsubchapters, areas that should be on the wiki but not in the book and links inside the document. That makes our whole embedded development team contribute to the Documentation, which is quite nice. I find this actually way nicer than doing LaTeX by hand or doing Docbook. When I see the Asteriskdoc Docbook files with their extraordinarily long lengths. ___ 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] Re: New Beta version of Grandstream Firmware 1.0.5.9
Can they tell whether it is a phone making the request? Yes, they can, because the phone sends TFTP extensions telling it's current firmware. The TFTP-Server then only sends new firmware if the firmware is actually new. ___ 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] Broadvoice - incoming calls problem
Broadvoice keeps changing servers, but instead of registering and sending your calls through sip.broadvoice.com , you now need to send them through 147.135.8.129 ... this is subject to change though. Call their main line if it comes about again. They have someone on call 24 hrs a day. ___ 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] 2 cards
What about outgoing How do I tell it all sales,sip 100+, to go out threw vpb card's channel and all admin,sip 200+ to go threw zaptel? Thanks for the help so far On Tue, 2004-07-27 at 16:59, Seth Remington wrote: On Tue, 2004-07-27 at 09:48, Altus Snyman wrote: Ya but the one is zaptel nd one voicetronix so it uses vpb.conf for example sales The vpb.conf file allows you to define contexts for each of the channels just like zapata.conf so there shouldn't be a problem. Just use one context in zapata.conf and a different one in vpb.conf. -Seth ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. A number of us are using SPA-3000s for this exact purpose, including myself. Works pretty well. -- PhoneBoy ___ 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] drivers, kernel 2.6 and distribution
On Tue, 2004-07-27 at 23:30, Florin Andrei wrote: Download the Asterisk 1.0-RC1 RPMs from here (the Fedora 1 packages since there are no Fedora 2 packages there yet): ftp://ftp.nacs.net/asterisk Well, download the FC1 SRPMs, because the binary FC1 RPMs are not ok on FC2. Unpack the zaptel src.rpm (rpm -ivh zaptel...src.rpm) Before that, rebuild the libpri src.rpm and install it. After installing the zaptel, the last one to rebuild and install is the asterisk package. -- Florin Andrei http://florin.myip.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] Re: New Beta version of Grandstream Firmware 1.0.5.9
Here is an example, made with tethereal -V host grandstream: User Datagram Protocol, Src Port: 32874 (32874), Dst Port: tftp (69) Source port: 32874 (32874) Destination port: tftp (69) Length: 393 Checksum: 0xd481 (correct) Trivial File Transfer Protocol Opcode: Read Request (1) Source File: bootload.bin Type: octet Option: blksize = 1024 Option: tsize = 0 Option: timeout = 4 Option: grandstream_MODEL = BT-100 Option: grandstream_NAT = 1 Option: grandstream_ID = 000b82013dc0 Option: grandstream_REV_BOOT = 001.000.000.018 Option: grandstream_REV_PHONE = 001.000.005.007 Option: grandstream_REV_VOC = 001.000.000.006 Option: grandstream_REV_HTML = 001.000.000.037 Option: grandstream_REV_RING1 = 001.000.000.000 Option: grandstream_REV_RING2 = 001.000.000.000 Option: grandstream_REV_RING3 = 000.000.000.000 ___ 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] problems with digium cvs????
Hi !! I hope you can help me. I can't connect to digium cvs: cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Connection refused. Arethere any problems on it? And how can I downloadit whitout cvs? Perhaps there are some others cvs sites? Thanks a lot for yours answers. Asmine
Re: [Asterisk-Users] Play CD!
I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. Make sure you are running mpg123 0.59r and no other version Jason ___ 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] Re: Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sjaakie Helderhorst Sent: Monday, July 26, 2004 5:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap' I got things running with ISDN4Linux See configuration example below, I found it exploring the WIKI-site. (to make an outgoing call users need to press 0*[number to call]) Hope this is useful. It seems an interesting solution, but I need echo cancellation and so I have to use zaphfc. Thanks, Alessandro ___ 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 unable to transfer
Dimitri Did you get a resolution to this problem? I am seeing the same, my * box talks to Telappliant using AIX, anybody else seen this? Roy.. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of reseaux Sent: 23 June 2004 10:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Iax unable to transfer Dear List I have notice this kind of problem between my two * box. My scenario is : Iax GSM IaxClient-PBX1PBX2--TDM today CVS Stable V1 I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join the two call i can see the log below from my PBX1, i can speak for few second and after the FireFly hangup. I have try to change * version from Stable to today CVS but no success same problem. I have enabled the IAX Debug and seems the RX side (PBX1) dont accept something from PBX2 and show the unable to transfer (im not expert) :-) The strange thing is if i call from Sip Phone/client i dont have a problem the Call is bridged! The events from the CLI: - Executing Dial([EMAIL PROTECTED]/5, IAX2/out:[EMAIL PROTECTED]/[EMAIL PROTECTED]|60|g) in new stack -- Called out:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 213.215.xx.xx (format GSM) -- Format for call is GSM -- IAX2[out]/6 stopped sounds -- IAX2[out]/6 is ringing -- IAX2[out]/6 stopped sounds -- IAX2[out]/6 answered [EMAIL PROTECTED]/5 -- Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[out]/6 -- Channel 'IAX2[out]/6' unable to transfer -- Hungup 'IAX2[out]/6' -- Executing Hangup([EMAIL PROTECTED]/5, ) in new stack == Spawn extension (incoming,001223445, 4) exited non-zero on 'IAX2 [EMAIL PROTECTED]/5' -- Executing Hangup([EMAIL PROTECTED]/5, ) in new stack --- Thanks in advance for possible help! Dimitri ___ 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] problems with digium cvs????
On Wed, 2004-07-28 at 03:04, Asmine Ouloube wrote: Hi !! I hope you can help me. I can't connect to digium cvs: cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Connection refused. Are there any problems on it? And how can I download it whitout cvs? Perhaps there are some others cvs sites? Check your DNS configuration. [EMAIL PROTECTED]:~$ host -v cvs.digium.com Trying cvs.digium.com ;; -HEADER- opcode: QUERY, status: NOERROR, id: 64064 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 2, ADDITIONAL: 2 ;; QUESTION SECTION: ;cvs.digium.com.IN A ;; ANSWER SECTION: cvs.digium.com. 86400 IN A 66.250.69.240 cvs.digium.com. 86400 IN A 66.225.202.81 ;; AUTHORITY SECTION: digium.com. 62912 IN NS marko.marko.net. digium.com. 62912 IN NS linux-support.net. ;; ADDITIONAL SECTION: marko.marko.net.62912 IN A 216.207.245.12 linux-support.net. 172799 IN A 216.207.245.1 Received 167 bytes from 192.168.123.254#53 in 284 ms -- 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
[Asterisk-Users] Optipoint 400 Standard Sip
Hello Wendys Steffen. I tried to change chan_sip.c that way, you told us. but my optiset isn´t working properly. may you be so kind enough to send me your configuration of an working optipoint ? Mine is working as Sip routing: GATEWAY. An outgoing call can be established, but due to not registering to asterisk an incoming call could not be delivered. HELP PLZ. ! ;-) Kind regards thx for help in advance Roland / Nuermberg / Germany P.S. Wendys können wir mal mails auf deutsch austauschen. wenn du schon aus nürnberg bist :-) ___ 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] Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, July 26, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap' Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the latest bri-stuff.0.1.0-RC2g (released just today!) and started the installation. I still have the problem! I really have no idea about what to do! Any suggestion would be greatly appreciated. Thanks, Alex ___ 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] IRC Etiquette
My comments on these matters is simple. We (newbies or experienced) still needs to learn from our experiences. Personally, I'm very appreciated when I asked a dumb question, someone replies me with the link to the documentation. Mostly it helps, but again, the documentation is not perfect. I think it's our job to make a perfect documentation so it will help others to understand more. Thanks Isianto ___ 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] BugetTone Bug Showstopper,
Hi! When TRANSFERING a call to another extension, if you enter an invalid extension, (I.e. Hit TRANSFER, then dial erroneous number.. SEND, Congestion tone, Hang-up, go off-hook.. Try different solutions to try to get call back But no.) Just don't use the GS transfer button. Instead use * transfer thru # or work with call parking (or valet parking aka bkw parking). Philipp ___ 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 unable to transfer using Telappliant
Update I have found that setting notransfer=yes enables me to call Telappliant numbers (09XX) and not get disconnected but if I call a BT number the call goes out via Telappliant to the BT phone, it rings, the client answers, they can hear you, but firefly does not know the other end has been answered and continues to ring, obviously then you cant hear the client. Is this a symptom of notransfer=yes or is there another problem? Firewall related maybe? Anyone with a working Telappliant account using IAX? Cheers! Roy -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roy Eddleston Sent: 28 July 2004 09:13 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Iax unable to transfer Dimitri Did you get a resolution to this problem? I am seeing the same, my * box talks to Telappliant using AIX, anybody else seen this? Roy.. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of reseaux Sent: 23 June 2004 10:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Iax unable to transfer Dear List I have notice this kind of problem between my two * box. My scenario is : Iax GSM IaxClient-PBX1PBX2--TDM today CVS Stable V1 I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join the two call i can see the log below from my PBX1, i can speak for few second and after the FireFly hangup. I have try to change * version from Stable to today CVS but no success same problem. I have enabled the IAX Debug and seems the RX side (PBX1) dont accept something from PBX2 and show the unable to transfer (im not expert) :-) The strange thing is if i call from Sip Phone/client i dont have a problem the Call is bridged! The events from the CLI: - Executing Dial([EMAIL PROTECTED]/5, IAX2/out:[EMAIL PROTECTED]/[EMAIL PROTECTED]|60|g) in new stack -- Called out:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 213.215.xx.xx (format GSM) -- Format for call is GSM -- IAX2[out]/6 stopped sounds -- IAX2[out]/6 is ringing -- IAX2[out]/6 stopped sounds -- IAX2[out]/6 answered [EMAIL PROTECTED]/5 -- Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[out]/6 -- Channel 'IAX2[out]/6' unable to transfer -- Hungup 'IAX2[out]/6' -- Executing Hangup([EMAIL PROTECTED]/5, ) in new stack == Spawn extension (incoming,001223445, 4) exited non-zero on 'IAX2 [EMAIL PROTECTED]/5' -- Executing Hangup([EMAIL PROTECTED]/5, ) in new stack --- Thanks in advance for possible help! Dimitri ___ 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] problems with digium cvs????
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 10:20 AM Subject: Re: [Asterisk-Users] problems with digium cvs On Wed, 2004-07-28 at 03:04, Asmine Ouloube wrote: Hi !! I hope you can help me. I can't connect to digium cvs: cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Connection refused. Are there any problems on it? And how can I download it whitout cvs? Perhaps there are some others cvs sites? Check your DNS configuration. [EMAIL PROTECTED]:~$ host -v cvs.digium.com Trying cvs.digium.com ;; -HEADER- opcode: QUERY, status: NOERROR, id: 64064 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 2, ADDITIONAL: 2 ;; QUESTION SECTION: ;cvs.digium.com.IN A ;; ANSWER SECTION: cvs.digium.com. 86400 IN A 66.250.69.240 cvs.digium.com. 86400 IN A 66.225.202.81 ;; AUTHORITY SECTION: digium.com. 62912 IN NS marko.marko.net. digium.com. 62912 IN NS linux-support.net. ;; ADDITIONAL SECTION: marko.marko.net.62912 IN A 216.207.245.12 linux-support.net. 172799 IN A 216.207.245.1 Received 167 bytes from 192.168.123.254#53 in 284 ms I check it : Trying cvs.digium.com ... Query done, 2 answers , status : no error The following answer is not authoritative: cvs.digium.com. 15698 IN A 66.250.69.240 cvs.digium.com. 15698 IN A 66.225.202.81 Authority information: digium.com. 15698 IN NS linux-support.net. digium.com. 15698 IN NS marko.marko.net. Additional information linux-support.net. 166099 IN A 216.207.245.1 marko.marko.net.17863 IN A 216.207.245.12 So I don't think that the problem come from my DNS. I try with cvsup but : cannot connect to cvs.digium.com: destination unreachable. ___ 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] Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
I still have the problem! I really have no idea about what to do! Any suggestion would be greatly appreciated. I once made a successful zaphfc setup (althought currently I'm on chan_capi). What I made has been put in python code into my setup tool. Maybe you try www.holgerschurig.de/destar.html and let it make a demo configuration for you. ___ 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] Polycom IP-600 leasing?
Hello, That is if you can find a company that will actually sell one to you. I have bought several IP600s in the past off of Froogle, but I tried to buy a new IP600 last month and 3 of the places listed on Froogle that were the cheapest said they couldn't sell the phone to non-authorized customers. Do you know of a company that will sell the IP600 at under $300 to a regular Asterisk user? MATT--- -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 1:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP-600 leasing? Um, these phones are less than $300 a piece. http://www.google.com/froogle?q=polycom+600scoring=psa=Nstart=10 Hard to find a leasing company for that small an amount, but I'm sure they're out there. John Scott Laird wrote: We're interested in leasing roughly 15 Polycom IP-600 phones. Does anyone have a vendor that they can recommend for this? 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 ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
Only for basic testing. By default, incoming pstn calls ring the fxs line. However, there is an option to disable that and apparently route the call to the voip system. There is apparently another option that involves a timeout, routing the call to * if the fxs doesn't answer within the timeout period. I've not played with those options as yet. I am most interested in using it for incoming calls. Have you tried that yet? /carmi On Jul 27, 2004, at 5:30 PM, Rich Adamson wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. In the process of doing that now. Simple / prelim implementation: Each of the three ports (eg, fxs, fxo, cat5) are treated as separate interfaces, and one can configure fxo - *, fxs - *, ring-through from fxo - fxs, * g/w functions to the pstn, etc. There seems to be a ton of functionality in the box and those functions are mostly limited by your imagination (and how well one can read and comprehend). Configurable from a web interface, however there are a ton of options that aren't very clear without digging deep into their newly released admin manual (called a user guide on their site). The manual seems to have been written for the 1000/2000 with additional chapters/sections oriented to the 3000. (Sort of rush to print.) The fxo and fxs interfaces can be configured to register separately with *, making both very addressable, etc. Like *, it also has an internal dialplan, however understanding the various interactions requires some experimentation, as each of the interfaces seem to be considered a gateway, and part of the dialplan directs calls to gw0, gw1, gw2 (etc) which correspond to physical interfaces in most cases. The box was truly targeted for the residential user where existing phones interface on one side, the pstn line on the other side, and the default call is sent to the voip interface. Disconnected (or failed) ethernet results in a relay flipping, tying the fxs directly to the fxo. Same with power failure. Nice. So, properly configured, it appears to be a very nice box that would allow * to sit in the middle, but still provide excellent fail-over capabilities when unusual events occur. For small installations, it makes handling US 911 calls extremely easy as that can be made part of the internal dialplan. Initial tests did not show any signs of echo, very good volume and audio quality, and would probably be a good choice for small quantities of pstn lines (particularily soho and residential users). The only downside I've seen thus far (not much experience as yet) is that * calls to the pstn line are cut through immediately, so one hears the initial dialtone from the pstn and the sending of the dtmf tones on all outgoing calls. Kind of annoying, but there might be some config option to handle it; I've just not found it as yet. (If anyone knows how to handle that, sure would appreciate a suggestion.) Thus far, I'd give the box at least an A-, and will likely move higher with a little more experience. Rich ___ 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 ---End of Original Message- ___ 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] Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, July 26, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap' Hi, The following is from Asterisk's log (asterisk -gc | tee asterisk.log): [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found Jul 28 12:34:29 WARNING[16384]: chan_skinny.c:2584 reload_conf ig: Unable to get our IP address, Skinny disabled == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Jul 28 12:34:29 WARNING[163850]: chan_oss.c:238 sound_thread : Read error on sound device: Resource temporarily unavailable Do you think that such warnigs may be somehow related to Unable to create channel of type 'Zap'? (Soundcard is an onboard VIA chipset based card) Thanks, Alex ___ 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] Shan:Help in configuring Dialplan
Hi, This is Shan. I need a suggestion regarding my work. First i'll explain my work and next my problem in the extensions.conf file: Work According to my dialplan every incoming DID's will be diverted to the main context. From the main context i've to send my incoming DID's to two different context. For example: 100 - 200 DID's will be diverted to context-1 and 300-400 will be diverted context-2. I've sepearte IVR features for context-1 context-2. Problem --- So here is my problem. How can i direct my calls to the IVR feature for context-1 context-2.All the calls comes to the main context initially and then i've to divert the calls to the appropriate context-1 or context-2. I dont' want to use include=context-1 or include=context-2 statements in the main context. Please give any other idea to complete this work. Regards Shan
[Asterisk-Users] IAX transfer bug in last CVS ?
I updated from CVS yesterday and today and still have the problem.
IaxComm cannot transfer the call when it's an outgoing call.
('outgoing' is from the dial plan point of view).
details : First I call the IaxComm phone and accept
the call. Then I'm not able to transfer it from the
IaxComm phone.
If the call is an incoming call it works fine.
details : First I call a phone from the IaxComm phone and
accept the call on the other phone. Then I'm able to
transfer the call from the IaxComm phone.
I saw that the manager api naming convention changed for IAX
channels (no more brackets). Any other change ?
I changed the dial string for my IAX phones. Instead of using
IAX2/peerip , I used IAX2/recordname.
(recordname is the name in brackets in the first line of the
phone entry in iax.conf)
With that change, one phone was fixed and the other was still
not able to transfer. I was not able to find the difference
between the two. Uh !
Any information ?
Thanks
___
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Re: [Asterisk-Users] debian install zaptel
Jonathan Moore wrote: I saw the same problem on a customer install and because of short time frame we wiped system and moved over to Fedora, since we already has tested and used it in the past. I actually had the opposite experience where fedora wasted crap loads of my time and wiped the hdds and I had asterisk on debian up and running in minutes... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ 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] shan:Needed help
Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial 10 after dialling 1234. Is it possible to do? Regards shan
[Asterisk-Users] Problems Compiling Asterisk-oh323-0.6.2
Hi. im compiling the wrapper for oh323(under Suse 9.0) -pwlib 1.6.6 -openh323 1.13.5. (with oh323 Patch) i execute: ./samples/simple/obj_linux_x86_r/simph323 and it works fine. When i Run asterisk-oh323 0.6.2: make I get the following errors: chan_oh323.c:660: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_oh323.c:660: error: initializer element is not constant chan_oh323.c:660: error: (near initialization for `oh323_ep_list.lock') make[1]: *** [chan_oh323.o] Fehler 1 make[1]: Leaving directory `/home/voip/Documents/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Fehler 1 Any ideas? Zayn ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. A number of us are using SPA-3000s for this exact purpose, including myself. Works pretty well. -- PhoneBoy Have you found a way to get rid of the dial tone and dtmf tones when placing an outbound pstn call through the 3000? In my config, the call completes as expected however the dialtone and dtmf tones are slightly annoying. Rich ___ 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] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
Anyone can comment this or just mailing is dead? Vasyl Rublyov wrote: I started to see this problem as soon as we connected to Verizon PRI (DMS-100 Switch) and it prints every 3-5 seconds. [Verizon DMS-100 PRI] [Lucent Merlin Legend] [Asterisk] Asterisk/LibPRI/Zaptel are built from HEAD CVS on Jul 10 2004. Any help? Jul 27 20:50:20 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) in pri debug: Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) Protocol Discriminator: Unknown (0) len=22 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) Protocol Discriminator: Unknown (0) len=22 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) Protocol Discriminator: Unknown (0) len=22 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) /etc/zaptel.conf: # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [channels] language=en context=default switchtype=national pridialplan=unknown overlapdial=no signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=32 echocancelwhenbridged=yes echotraining=yes rxgain=1.5 txgain=5.5 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no musiconhold=default channel = 1-23 ___ 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] shan:Needed help
Shanmuganathan Kumaravel wrote: Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial 10 after dialling 1234. Is it possible to do? Shan, Welcome to the Asterisk community! The questions you ask indicate that you need to do your homework. These are basic functions, so please read the available documentation. Some suggestions on where to start: * Asterisk: http://www.asterisk.org * Asterisk mailing lists: http://lists.digium.com (users, bsd, dev, biz and cvs mailing list) * Asterisk bug tracker: http://bugs.digium.com * Asterisk IRC channel: #asterisk on irc.freenode.net * Digium: http://www.digium.com * Wiki: http://www.voip-info.org * Voip Search: http://search.voip-forum.com * Astricon 2004: http://www.astricon.net * Asterisk documentation project: http://www.asteriskdocs.org Also, the IRC channel give online-help and advice. Best regards, /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
Re: [Asterisk-Users] shan:Needed help
On 28 Jul 2004 11:27:24 -, Shanmuganathan Kumaravel [EMAIL PROTECTED] wrote: I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial 10 after dialling 1234. Is it possible to do? Yes this is possible. I suggest taking a read of chapter 4 at http://www.asteriskdocs.org (Introduction to Dialplans) Thanks, Leif Madsen. http://www.asteriskdocs.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] Iax unable to transfer
On Wed, 2004-07-28 at 09:12 +0100, Roy Eddleston wrote: Dimitri Did you get a resolution to this problem? I am seeing the same, my * box talks to Telappliant using AIX, anybody else seen this? I don't know if it's exactly the same thing but I had lots of trouble with two *s trying to communicate using Firefly at both ends, but with a GS-BT101 at one end no problem. Exactly as described Hello Fred, this is cut off. As I've got Grandstreams and SNOMs at both ends now I have not tried to find the solution as no one wanted soft phones anyway. -- Dave Cotton [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] shan:Needed help
Please don't use HTML e-mail. I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial quot;10quot; after dialling quot;1234quot;. Is it possible to do? I use my Grandstream in Early-Dial-Mode and here it is possible. For a better answer, I need to know the relevant part of your extension.conf. ___ 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] Can't dial SIP-EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Jul 28 12:34:29 WARNING[16384]: chan_skinny.c:2584 reload_conf ig: Unable to get our IP address, Skinny disabled == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Jul 28 12:34:29 WARNING[163850]: chan_oss.c:238 sound_thread : Read error on sound device: Resource temporarily unavailable Do you think that such warnigs may be somehow related to Unable to create channel of type 'Zap'? (Soundcard is an onboard VIA chipset based card) I don't think so. However, I only load as little modules into asterisk at possible, e.g. my modules.conf starts with a autoload=no. However, the Unable to get our IP address looks fishy anyway. Enter the command hostname in your shell and make sure that this domainname points to your IP into /etc/hosts. Afterwards, you should be able to do a ping `hostname` without any errors. ___ 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] Outgoing works, incoming doesn't...
Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/usernameHostDyn Nat ACL Mask Port Status 105/105 192.168.2.175D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the config on that phone? If so, who can tell me what? Regards, Evert ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
I found it was worse when using the G726 or G723 codecs, but if you used the G711 codec, the DTMF echo was hardly noticable. I was using the latest image: 2.0.9d -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Wednesday, July 28, 2004 8:31 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *? On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. A number of us are using SPA-3000s for this exact purpose, including myself. Works pretty well. -- PhoneBoy Have you found a way to get rid of the dial tone and dtmf tones when placing an outbound pstn call through the 3000? In my config, the call completes as expected however the dialtone and dtmf tones are slightly annoying. Rich ___ 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: problems with digium cvs????
In article [EMAIL PROTECTED], Asmine Ouloube [EMAIL PROTECTED] wrote: On Wed, 2004-07-28 at 03:04, Asmine Ouloube wrote: Hi !! I hope you can help me. I can't connect to digium cvs: cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Connection refused. [...] I check it : Trying cvs.digium.com ... Query done, 2 answers , status : no error The following answer is not authoritative: cvs.digium.com. 15698 IN A 66.250.69.240 cvs.digium.com. 15698 IN A 66.225.202.81 Authority information: digium.com. 15698 IN NS linux-support.net. digium.com. 15698 IN NS marko.marko.net. Additional information linux-support.net. 166099 IN A 216.207.245.1 marko.marko.net.17863 IN A 216.207.245.12 So I don't think that the problem come from my DNS. I try with cvsup but : cannot connect to cvs.digium.com: destination unreachable. Have you got an entry for cvs.digium.com in your /etc/hosts file? Perhaps you put one there in the past when having problems. If so, remove it again! Cheers Tony -- 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
[Asterisk-Users] Access voicemail from Cisco 7960
Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? Regards, Evert ___ 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] BugetTone Bug Showstopper,
On Wed, 28 Jul 2004, Philipp von Klitzing wrote: Hi! When TRANSFERING a call to another extension, if you enter an invalid extension, (I.e. Hit TRANSFER, then dial erroneous number.. SEND, Congestion tone, Hang-up, go off-hook.. Try different solutions to try to get call back But no.) Just don't use the GS transfer button. Instead use * transfer thru # or work with call parking (or valet parking aka bkw parking). Valet Parking is now known as simply Valet to prevent people from confusing Call Parking with it. This was done per Kram's request, so please, let's start calling it by it's new name. ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Commercial Asterisk Support
Hi there, I'm wanting to source some commercial support for the setup of a series of Asterisk Boxes to work with both H323 and SIP. Could people please contact me off-list that are proficient in full setups of Asterisk with H323/SIP Support for commercial purposes ? Cheers, Sahil ___ 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] Asterisk voicemail from mysql no longer working
Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying No entry in voicemail config file for 'number' whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though I can see that it is enabled and logs in when asterisk starts. I am sure I am missing something very basic, but could not find what ! Please help. ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.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
Re: [Asterisk-Users] Asterisk voicemail from mysql no longer working
Further invetigation revealed that app voicemail did not like the fact that I had the context set to 'local' as apposed to 'default' Any ideas' or shall I raise this as a bug ? Umar. --- Umar Sear [EMAIL PROTECTED] wrote: Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying No entry in voicemail config file for 'number' whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though I can see that it is enabled and logs in when asterisk starts. I am sure I am missing something very basic, but could not find what ! Please help. ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.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 ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.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] MGCP Caller ID
Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display Asterisk as the cid name, nothing for number. Here's what I have in my mgcp.conf for the device: [2084728800103] host = dynamic context = westcomllc line = aaln/1 callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] When placing outbound calls (out our pstn gateway), I always replace cid name/number w/the main number name of the company, so that direction it's not an issue -- just internal calls. Anyone seen this have ideas about what to do with it? Thanks, Jeremy Jones ___ 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] Best Linux for Asterisk
Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? Andy --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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] Outgoing works, incoming doesn't...
Evert Meulie wrote: Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/usernameHostDyn Nat ACL Mask Port Status 105/105 192.168.2.175D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the config on that phone? If so, who can tell me what? As Asterisk tells you, it's UNREACHABLE from Asterisk. Turn on SIP debug and see what happens - where Asterisk is sending packets and if we get any replies at all. /O ___ 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] Trouble compiling asterisk-addons MySQL
Title: Trouble compiling asterisk-addons MySQL Hi All, I am having trouble compiling the mysql addon for asterisk. I had downloaded the most recent version from CVS and placed it in /usr/src/ and I get the following error below. [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done BTW. I have asterisk running just fine. Thanks, James
RE: [Asterisk-Users] Best Linux for Asterisk
Hi Andy, I have had tremendous success running Asterisk on Slackware linux version 9.1. Its very quick to install and I had absolutely no problem compiling the source code for Asterisk or anything else so far. I have asterisk running on 2 servers right now that use Slackware. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Kirkland Sent: Wednesday, July 28, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux for Asterisk Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? Andy --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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] MGCP Caller ID
Try this: mgcp.conf [2084728800103]host = dynamiccontext = westcomllccallerid = "Jeremy Jones" 103nat = notransfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED] line = aaln/1 - Original Message - From: Jeremy Jones To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:13 AM Subject: [Asterisk-Users] MGCP Caller ID Good Morning,I'm having an issue with callerid display when calles are placed _from_an mgcp device (8x8 ata w/mgcp firmware). Internally, there are severaldifferent sip devices and one mgcp device. Calls from any of the sipdevices to any other device (sip or mgcp) have name/number displayedproperly by the called party's phone. Calls from the mgcp device to anyother device display "Asterisk" as the cid name, nothing for number.Here's what I have in my mgcp.conf for the device:[2084728800103]host = dynamiccontext = westcomllcline = aaln/1callerid = "Jeremy Jones" 103nat = notransfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED]When placing outbound calls (out our pstn gateway), I always replace cidname/number w/the main number name of the company, so that directionit's not an issue -- just internal calls. Anyone seen this have ideas about what to do with it?Thanks,Jeremy Jones___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
Most die hard Linux fans build their own distribution. But there is also gentoo which is VERY popular. - Original Message - From: Eric Kirkland To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:13 AM Subject: [Asterisk-Users] Best Linux for Asterisk Hi folks; Can anyone recommend the best Linux OS (versions, etc) to runAsterisk? I'd like to be able to run the Text To Speech apps and some ofthe extended functions of the software (no phone hardware needed, all Voiceover IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion10 I think?) but I'm having difficulty compiling the TTS stuff.I'm just wondering if there's a widely used version that pretty much workswith everything...?Andy---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Linux for Asterisk
Oh boy, time for distro wars :) I have found that for some reason Asterisk seems to run better on Slackware than Redhat, that's just my personal non-scientific observations, but we have 4 Asterisk servers in production(two redhat 9, one slackware 9.1 and one slackware 10.0) with almost identical hardware and the Slackware boxes have a lower average load over the same Asterisk usage. I have also talked with several people who are very happy with Mandrake, Gentoo and Debian. Those are the distros that most of the Asterisk crowd seem to use. There is also an Asterisk-centric distro but I haven't heard much about it lately. And as always check out the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+linux+distributions MATT--- -Original Message- From: Eric Kirkland [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux for Asterisk Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? Andy --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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] Best Linux for Asterisk
On Wed, 28 Jul 2004 09:13:37 -0400, Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? I personally use Fedora Core 1 and 2 successfully at home. Gentoo seems to be the most widely agreed upon distribution though. I don't think anyone would slam you for using Asterisk on it. HTH, Leif Madsen. http://www.asteriskdocs.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] Outgoing works, incoming doesn't...
Hmm, I get lots of these: to 192.168.2.175:5060 Retransmitting #3 (no NAT): OPTIONS sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.6:5060;branch=z9hG4bK442bde8b From: asterisk sip:[EMAIL PROTECTED];tag=as6496d70e To: sip:192.168.2.175 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 28 Jul 2004 13:44:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 IP 192.168.2.175 is the phone IP 192.168.11.6 is Asterisk (it's not a routing problem, since other phones on the 192.168.2.x IP's do show up as 'OK') Regards, Evert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: 28 July 2004 15:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outgoing works, incoming doesn't... Evert Meulie wrote: Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/usernameHostDyn Nat ACL Mask Port Status 105/105 192.168.2.175D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the config on that phone? If so, who can tell me what? As Asterisk tells you, it's UNREACHABLE from Asterisk. Turn on SIP debug and see what happens - where Asterisk is sending packets and if we get any replies at all. /O ___ 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] Best Linux for Asterisk
Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? Best is highly subjective, and asking is likely to provoke a holy war ;) I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. Yes, I found that the TTS resisted my attempts to make it work on a Slackware box. However, it works fine on Debian 3.0 (which is also my preferred Linux distribution.) -- [About a discussion of heavily customised cars.] I thought they were talking about cheap whores - smelly, ugly, brightly coloured, waste of money, and got a cock inside them most of the time. -- Will Hargrave in uknot ___ 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] Rate Engine Compile Error
I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an Error 1 after typing make but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly Error 1 Anybody successfully compile Rate Engine? The least cost routing module for Asterisk? Thanks in advance. - [EMAIL PROTECTED] rate-engine]# make cc -O3 -W -Wall -Wmissing-prototypes -Wstrict-prototypes -Wshadow -g -fno-in line-functions -D_REENTRANT -I/usr/include/pcre -DWITH_MYSQL -I/usr/include/ mysql -c -o rate_engine.o rate_engine.c rate_engine.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) rate_engine.c: In function `cdr_ratecall': rate_engine.c:450: warning: implicit declaration of function `ast_channel_walk' rate_engine.c:450: warning: assignment makes pointer from integer without a cast rate_engine.c:450: warning: assignment makes pointer from integer without a cast rate_engine.c: In function `poster_worker': rate_engine.c:652: warning: unused parameter `arg' rate_engine.c: In function `rates_reload': rate_engine.c:1801: warning: unused parameter `argc' rate_engine.c:1801: warning: unused parameter `argv' rate_engine.c: In function `rates_status': rate_engine.c:1813: warning: unused parameter `argc' rate_engine.c:1813: warning: unused parameter `argv' make: *** [rate_engine.o] Error 1 [EMAIL PROTECTED] rate-engine]# - ___ 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] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
Dear Vasyl I have a E100P on my Asterisk Box connect to Definity G3 and i run Asterisk CVS-HEAD-07/19/04-13:47:15 I dont see this kind of mex as you say but every minute i have this mex: Jul 28 15:26:40 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:26:44 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:27:28 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 Jul 28 15:27:28 NOTICE[229390]: chan_zap.c:7001 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 --- But everything seems to work great in this box i dont have a very load use. Bye Dimitri On Wednesday 28 July 2004 01:41 pm, Vasyl Rublyov wrote: Anyone can comment this or just mailing is dead? Vasyl Rublyov wrote: I started to see this problem as soon as we connected to Verizon PRI (DMS-100 Switch) and it prints every 3-5 seconds. [Verizon DMS-100 PRI] [Lucent Merlin Legend] [Asterisk] Asterisk/LibPRI/Zaptel are built from HEAD CVS on Jul 10 2004. Any help? Jul 27 20:50:20 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) in pri debug: Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:23 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) Protocol Discriminator: Unknown (0) len=22 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:25 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) Protocol Discriminator: Unknown (0) len=22 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:27 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) Protocol Discriminator: Unknown (0) len=22 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: ALERTING (1) [97] Locking Shift (len=01): Requested codeset 7 [41 0e 02 01 21 54 01 21 54 01 21 54 01 21 54 00] Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: !! Unknown IE 1857 (len = 16) Jul 27 20:30:29 WARNING[98310]: chan_zap.c:6665 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0) /etc/zaptel.conf: # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [channels] language=en context=default switchtype=national pridialplan=unknown overlapdial=no signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=32 echocancelwhenbridged=yes echotraining=yes rxgain=1.5 txgain=5.5 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no musiconhold=default channel = 1-23 ___ 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] Trouble compiling asterisk-addons MySQL
Title: Trouble compiling asterisk-addons MySQL Did you check whether mysql client libraries and headers were installed? -Kannaiyan http://www.goods2world.com- Your VoIP Shop - Original Message - From: James Freire To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 2:23 PM Subject: [Asterisk-Users] Trouble compiling asterisk-addons MySQL Hi All, I am having trouble compiling the mysql addon for asterisk. I had downloaded the most recent version from CVS and placed it in /usr/src/ and I get the following error below. [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done BTW. I have asterisk running just fine. Thanks, James
RE: [Asterisk-Users] Best Linux for Asterisk
Andy, I have limited Linux experience, but I have Asterisk installed on a Slackware 10 box. I had some assistance from our developer (he knows a lot more than I do with using Linux). It seemed to install quick and I don't recall having any problems. ___ 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] Trouble compiling asterisk-addons MySQL
Hello James, Wednesday, July 28, 2004, 5:23:16 PM, you wrote: JF [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install JF ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` JF cdr_addon_mysql.c:33:19: mysql.h: No such file or directory JF cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory JF for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done You need to install libmysqlclient-devel (or alike) package with relevant header files. -- Best regards, Olegmailto:[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] MGCP Caller ID
Hi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Wednesday, July 28, 2004 7:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP Caller ID Try this: mgcp.conf [2084728800103] host = dynamic context = westcomllc callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1 Aha! Yup, that did the trick. So order matters there... Thanks, jeremy ___ 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] Trouble compiling asterisk-addons MySQL
Well. I have seemed to get a little farther with the problem. I added in a line in to the Makefile of CFLAGS+=-I/usr/local/mysql/include/mysql Now I get an error that has to do with mysqlclient below... I have also included my entire Makefile below the error. Thanks [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include`ls *.c` cc -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-linux/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 # # Asterisk -- A telephony toolkit for Linux. # # Makefile for CDR backends (dynamically loaded) # # Copyright (C) 1999, Mark Spencer # # Mark Spencer [EMAIL PROTECTED] # # This program is free software, distributed under the terms of # the GNU General Public License # MODS= CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I/usr/local/mysql/include/mysql CFLAGS+=-D_GNU_SOURCE INSTALL=install INSTALL_PREFIX= ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk MODULES_DIR=$(ASTLIBDIR)/modules # # MySQL stuff... Autoconf anyone?? # MODS+=$(shell if [ -d /usr/local/mysql/include/mysql ] || [ -d /usr/include/mysql ] || [ -d /usr /local/include/mysql ] || [ -d /opt/mysql/include ]; then echo cdr_addon_mysql.so; fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include/mysql ]; then echo -I/usr/local/mysql/include ; fi) CFLAGS+=$(shell if [ -d /usr/include/mysql ]; then echo -I/usr/include/mysql; fi) CFLAGS+=$(shell if [ -d /usr/local/include/mysql ]; then echo -I/usr/local/include/mysql; fi) CFLAGS+=$(shell if [ -d /opt/mysql/include/mysql ]; then echo -I/opt/mysql/include/mysql; fi) MLFLAGS= MLFLAGS+=$(shell if [ -d /usr/lib/mysql ]; then echo -L/usr/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /usr/local/mysql/lib ]; then echo -L/usr/local/mysql/lib; fi) MLFLAGS+=$(shell if [ -d /usr/local/lib/mysql ]; then echo -L/usr/local/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /opt/mysql/lib/mysql ]; then echo -L/opt/mysql/lib/mysql; fi) all: depend $(MODS) install: all for x in $(MODS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done clean: rm -f *.so *.o .depend %.so : %.o $(CC) -shared -Xlinker -x -o $@ $ ifneq ($(wildcard .depend),) include .depend endif cdr_addon_mysql.so: cdr_addon_mysql.o $(CC) -shared -Xlinker -x -o $@ $ -lmysqlclient -lz $(MLFLAGS) depend: .depend .depend: ./mkdep $(CFLAGS) `ls *.c` -Original Message- From: Oleg A. Arkhangelsky [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 9:48 AM To: [EMAIL PROTECTED]; James Freire Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Trouble compiling asterisk-addons MySQL Hello James, Wednesday, July 28, 2004, 5:23:16 PM, you wrote: JF [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install JF ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` JF cdr_addon_mysql.c:33:19: mysql.h: No such file or directory JF cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory JF for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done You need to install libmysqlclient-devel (or alike) package with relevant header files. -- Best regards, Olegmailto:[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] 2 cards
On Wed, 2004-07-28 at 03:07, Altus Snyman wrote:
What about outgoing
How do I tell it all sales,sip 100+, to go out threw vpb card's channel
and all admin,sip 200+ to go threw zaptel?
Thanks for the help so far
You also do this through contexts. In your sip.conf you assign a context
to each phone like so:
[sip100]
type=friend
username=sip100
secret=XXX
callerid=Buckaroo Bonzai 100
host=dynamic
context=sales--- /* right here */
[EMAIL PROTECTED]
Then in your extentions.conf you would have:
[sales]
ignorepat = 9
exten = _91NXXNXX,1,Dial(Zap/1/${EXTEN:1}) /* use the zap */
exten = _91NXXNXX,2,Congestion
The above example would allow the sales people to dial 9 to dial a long
distance number on the Zaptel card. The context you use for admin would
explicitly use the vpb card instead.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Re: [Asterisk-Users] Problems Compiling Asterisk-oh323-0.6.2
Zineddin Karzazi wrote: Hi. im compiling the wrapper for oh323(under Suse 9.0) -pwlib 1.6.6 -openh323 1.13.5. (with oh323 Patch) i execute: ./samples/simple/obj_linux_x86_r/simph323 and it works fine. When i Run asterisk-oh323 0.6.2: make Download and install version 0.6.3a. Michael. ___ 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] Best Linux for Asterisk
Have there been noticed any differences in echo from distro to distro on the very same hardware? I mean install a distro compile and run *, then replace it with another distro on the same box and cards. That could be intersting. Thanks, Yiannis. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: 28 July 2004 14:29 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best Linux for Asterisk On Wed, 28 Jul 2004 09:13:37 -0400, Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? I personally use Fedora Core 1 and 2 successfully at home. Gentoo seems to be the most widely agreed upon distribution though. I don't think anyone would slam you for using Asterisk on it. HTH, Leif Madsen. http://www.asteriskdocs.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 ___ 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] Best Linux for Asterisk
On Wed, 2004-07-28 at 09:13, Eric Kirkland wrote: I'm having difficulty compiling the TTS stuff. You aren't very specific about the problems you are having compiling Festival but on the off chance that your problems were the same ones I had you might want to check out this: http://sremington.zapto.org/weblog/2004-07-04_14.52.21.html -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] converting gsm file to g729 format
hi all; Anybodey know how to convert astersik *.gsm files (like voicemail ivr prompts) to .g729 format? Regards mohammad
Re: [Asterisk-Users] RE: Integrated Networks IN1002 SIP Phones?
Did you buy from this guy Michael Wang? I used to buy Cisco ATAs from him. Nice guy to deal with. Back to the phones. From the pre-release docs I got from michael, I gather that the IN100x phones and IW1688 gateway are all based on VoIP SoC (system on chip) from Centrality http://www.centrality.com.cn/solutions/productdocuments.htm. The PA1688 SoC is very common in VoIP phones made in China/Taiwan. I suppose that should provide you a large source of 'compatible' phones. FYI. Randy MacKay wrote: I would like to order more phones, but I'm concerned about the history of the phones and the company. The post on ebay said they are the manufacture, and my emails have always been answered. Their is no working link on the website to download new firmware, but they say they will email me a copy when the next version is available. Updates are suppose to happen aprox every 3-4 weeks. My beginners assessment of the IN1002 SIP Phones: I was able to configure the Integrated Networks IN1002 SIP Phones from the web interface. The PDF Manual was adequate enough for me to get it working with Asterisk. I like the feel and display a little better than the Grandstream, but its not a Cisco. The price was very affordable ($106.99 w/ next day shipping from China) and larger discounts on 10 or more phones. Ring tone is changeable. Speaker Voice readout for IP address. Button for list of missed calls, but no time stamp to distinguish between new and old calls. Ring tone in ear set on SIP/SIP calls I was not able to change, and it almost sounds like a busy signal. Ring tone in ear set to Pots line, ring tone was normal. So far I have not figured it out. My concerns: No message waiting light/message No missed calls message you have to push the missed calls button, but can t tell when it was missed Will I be able to get support / parts in a few years? Randy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Monday, July 26, 2004 3:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Integrated Networks IN1002 SIP Phones? Hi! Does anyone know anything about the Integrated Networks IN1002 SIP Phones from China? I ordered one of these phone off of ebay and I have it working. I was wondering if anyone else knows anything about them? Their website www.integratednetworks.com.cn was not very helpful, and the online PDF manual is not OK. I don't have such a phone, but: Please describe your problem and/or the info that you are looking for, including what works with * and what doesn't. That'll be of help for everyone. Cheers, Philipp --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.725 / Virus Database: 480 - Release Date: 7/19/2004 ___ 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] Best Linux for Asterisk
-Original Message- From: Eric Kirkland [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux for Asterisk Well, I am not going to say what is or isn't the best. (Because everything is good depending on skill level and or circumstances.) I personally have three asterisk boxes running on Gentoo 2004.1 with great success. Hope this helps, Robert Jackson ___ 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] Rate Engine Compile Error
Deon Rodden wrote (on Jul 28): I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an Error 1 after typing make but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly Error 1 There's a clue in the line that says error near the start... rate_engine.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) It needs updating to use a new version of the AST mutex definition macro. It's pretty trivial if you look at similar lines in the asterisk source. Chris. -- == [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] Best Linux for Asterisk
It may sound bad, but I use Fedora Core 1. However, I installed using reiserfs (my preferred filesystem) and I installed all the updates and had to custom compile a new kernel (as the stock one that comes with Fedora is too screwy, and the sources aren't done right and certain programs wouldn't compile). However, once I did the updates and used a custom compiled kernel, everything runs fast/smooth on it. I haven't had an issue compiling anything Asterisk related, yet. I did have issues with my Redhat 9 server, and my OpenNA 1.0 server. Of course, I have put a lot of security and performance tweaks into my Fedora installation, had to make it less redhat but now it runs good. You may not be willing to do all that work; but if you're familiar with redhat and how they do things and their directory structures and common commands, I think Fedora Core 1 is a good choice. Just a FYI. I think you'll find that as long as your favorite distro is decent, Asterisk will work. - Original Message - From: Peter Corlett [EMAIL PROTECTED] Newsgroups: newsgate.asterisk-users To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 9:29 AM Subject: Re: [Asterisk-Users] Best Linux for Asterisk Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? Best is highly subjective, and asking is likely to provoke a holy war ;) I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. Yes, I found that the TTS resisted my attempts to make it work on a Slackware box. However, it works fine on Debian 3.0 (which is also my preferred Linux distribution.) -- [About a discussion of heavily customised cars.] I thought they were talking about cheap whores - smelly, ugly, brightly coloured, waste of money, and got a cock inside them most of the time. -- Will Hargrave in uknot ___ 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] Best Linux for Asterisk
Peter Corlett [EMAIL PROTECTED] wrote: Eric Kirkland [EMAIL PROTECTED] wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? Best is highly subjective, and asking is likely to provoke a holy war ;) Agreed. Having said that, Gentoo is clearly the best. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] Display and UUS IEs on PRI - Q.931 question
Hi Klaus, Thank you very much for your pointers. I applied your patches and uncommented and changed line 8007 to look like this: ast_sendtext(pri-pvts[chanpos]-owner, e-display.text); instead of ast_sendtext(pri-pvt[chan]-owner, e-display.text); Unfortunately the display IEs still don't seem to show up in asterisk (they *do* show up in the pri debug output). Any ideas why? Best regards martin Klaus-Peter Junghanns wrote: Hi, you can take a look at how bristuff does this (it only has to be enabled in chan_zap to actually forward the display IE, uncomment line 8007). Latest version of bristuff is 0.1.0-RC2g which works with todays CVS versions. You can find it at www.junghanns.net/asterisk/ best regards Klaus -- Martin A. Blatter | lic. oec. publ. Wirtschaftsinformatiker | IT-Leiter OLMeRO AG | Europastrasse 30 | CH-8152 Glattbrugg | Switzerland [EMAIL PROTECTED] | phone +41 44 200 44 50 ___ 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] Pickup zap channel already in use?
I'm getting a ton of noise on the channel just from the * side when I pickup my zap channel. Otherwise it works fine, if the other person in the house hangs up the noise goes away.. On Tue, 27 Jul 2004 23:30:08 -0400, Mark Woods [EMAIL PROTECTED] wrote: Kent wrote: On Tue, 27 Jul 2004 16:58:47 +, Mark Woods [EMAIL PROTECTED] wrote: I have to admit that your question interested me because I'm thinking of setting up the same thing. As of yet, though, I haven't found an answer to it. It's fairly simple when * has picked up, but I haven't really devoted much time to figuring out how to do it when it hasn't. So...let me work on it, and I'll let you know what I come up with. It's going to take at least a week, though, as I'm going to Oshkosh for the EAA Airventure on Thursday...but I'll see what I can come up with after that. -Mark A friend of mine who is another * user suggested using an extension with an empty Dial statement to connect my sip phone to the zap channel. I am going to try that tonight and see if that works. Let me know if you figure out anything else. Thanks! I actually got a chance to try it just now. Works like a champ! It has the added benefit of giving direct access, with dialtone, to the outside line, instead of having * dial. Here's what I put in my extensions.conf: exten = 4000,1,Dial(Zap/1/) exten = 4000,2,Congestion -Mark ___ 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] Desired Install in MotorHome
I've got a client who would love to be able to leave an asterisk server running sompelace, and achieve telephone connectivity using an IP phone from within his Motorhome in his words I want to be able to work from a mountaintop in Glacier National Park I've done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as completely un-workable. I understand that this is an extreamly hostile environment, the satelite uplink itself introduces too much latency for a standard configuration to work (1500ms) which is most likely where the problems come from. What I'm wondering is if anyone has ever succeded in making a setup like this work. Different protocol (H.323 MCGP etc) or different codecs? Our testing in his driveway revealed the following. 1. Incoming calls achieve a ring almost immediately, when answered there is 1 to 2 seconds delay in the conversation (like a really poor trans-atlantic call) 2. Outgoing calls fail... the phone returns Proxy Authentication Required however a few seconds later when the handset is picked up the inbound leg of the RTP stream is preseant on the phone. Outgoing audio is non-existant 3. Attempts to disconnect from that audio stream fail, the * server is simply not seeing the hangup from the phone. So whats everyones opinion, worth exploring further, or am I wasting my time trying? Thanks in advance. Paul M. Oster ___ 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] only one call at the time
I need to made an only one call to certain Ip using h323 at a time , an recive only one call a a time, is it possible to made ?
Re: [Asterisk-Users] Polycom IP-600 leasing?
On Jul 27, 2004, at 10:14 PM, John Baker wrote: Um, these phones are less than $300 a piece. http://www.google.com/froogle?q=polycom+600scoring=psa=Nstart=10 Hard to find a leasing company for that small an amount, but I'm sure they're out there. John They're that low? I've hard a hard time finding reasonable-looking vendors for under $335, and most places seemed closer to $400. Anyway, the issue is that I'm looking for 15 of them, which is a big enough total that management'd rather not write the check for them all at once. The joys of startups. It's been a while since I've had to be this cash-frugal, so I'm kinda rusty. 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] Rate Engine Compile Error
Deon Rodden [EMAIL PROTECTED] wrote: I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an Error 1 after typing make but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly Error 1 rate_engine.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) The above doesn't look like a standard/non-critical warning to me. Perhaps there's an update for the source you have. I don't have that code myself, so I don't know. You could try contacting the author. If you have no joy there then search the code for lines like the following: static ast_mutex_t foo = AST_MUTEX_INITIALIZER; and change to: AST_MUTEX_DEFINE_STATIC(foo); I.e. do as the error message suggests and use AST_MUTEX_DEFINE_STATIC rather than AST_MUTEX_INITIALIZER. You should also correct those other standard/non-critical warnings while you're in there. If the code is GPLed then please submit your changes back to the author for inclusion in the next release. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] broadvoice/asterisk
I've got srvlookup=yes, insecure=yes, and an entry in /etc/hosts for 147.135.8.128. Registration is fine, however if an incoming call (from broadvoice) arrives from 147.135.8.129, the call fails. So I added a sip.conf entry like: [sip-broadvoice] type=user context=from-broadvoice deny=0.0.0.0/0.0.0.0 permit=147.135.8.129/255.255.255.0 permit=147.135.0.129/255.255.255.0 which seems to correct the problems with broadvoice calls arriving from different broadvoice servers. Anyone see an issue with this approach, or, is there a better way to handle this? Rich Also make sure that you have insecure=yes in your friend/peer section of you sip.conf file. Sorry forgot to mention. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: Tuesday, July 27, 2004 3:20 AM To: Asterisk User (E-mail) Subject: [Asterisk-Users] broadvoice/asterisk Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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 ---End of Original Message- ___ 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] Problems connecting xlite phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Geoff Nordli wrote: | How simple it is to kiss a couple of days away over something really minute. | | It was definitely a client configuration issue. I configured Proxy 1 to | attach to asterisk. I really needed to configure [Default]. Once I | configured Default then I was off to the races. Or I think you might be able to right-click on the interface (which doesn't work under Wine, last time I checked; it locks up the interface) and choose the account you want to use. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFBB7y/uYsUrHkpYtARAh4kAJ4kn9qD3hitrOiEL+g3BZ7GtPz84wCcD2wz poAvcLAHq6y/LMoNJWp2T0M= =9bTl -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ 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] Access voicemail from Cisco 7960
On Wed, 28 Jul 2004 14:22:17 +0200, Evert Meulie [EMAIL PROTECTED] wrote: Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? I had a similar problem. If you look at the console, you'll probably either see it missing digits, or sending too many digits. Even though I was using ulaw as my codec, Asterisk didn't like my specifying dtmfmode=inband. I commented out that line and away it went fine. Here is my current sip.conf for my 7960 which works (connected directly to the Asterisk box) [100] type=friend secret=password username=100 callerid=Leif Madsen 18924 context=extensions ;dtmfmode=inband qualify=yes nat=no host=dynamic canreinvite=no disallow=all allow=ulaw allow=alaw allow=g729 mailbox=100 HTH, Leif Madsen. http://www.asteriskdocs.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
[Asterisk-Users] Asterisk-and-MacOSX News
A brief update on a few things which should be interesting (Bto anybody who's got a Mac running OSX ... (B (B (B1) JPT, integrated desktop dialer now with Asterisk (Bsupport (and X-Lite too) (B (BJon Nathan, the author of Jon's Phone Tool (JPT) has just (Breleased version 2.0.4 which supports dialing through (Blocal and remote Asterisk servers. It also supports (Bdialing through a locally installed X-lite. (B (BIt comes with plug-ins for major desktop apps, so you can (Bdial out through Asterisk directly from your OSX (BAddressbook or Microsoft Entourage etc. (B (BI have been playing with the pre-release for the last week (Band found it to be a really nice app. Check it out (B (Bsee screenshots of Asterisk related stuff here ... (B (Bhttp://www.voip-info.org/tiki-index.php?page=JPT (B (Bfor more info and to download, go directly to the author's (Bweb site here ... (B (Bhttp://homepage.mac.com/jonn8/jpt (B (B (B2) Asterisk 1.0 RC1 built and running on OSX 10.3.4 (B (BWe have now got Asterisk 1.0 RC1 running on OSX 10.3.4 (B"Panther" (it doesn't build on 10.2 "Jaguar" yet) and so (Bfar it seems to work. (B (BI am working on a new OSX install package for this and (Bhope to be able to release the package before the end of (Bthe coming weekend. (B (BThis will also include an Aqua GUI assistant to add new (Bextensions and phones, further I am planning to complete (Ban Assistant for configuring FWD and Voicepulse. (B (BAny service providers who'd like to get their own (BAssistant, or who'd be interested to throw a trial account (Binto the Assistant or other promotional requests, please (Bcontact me off-llist at benjamin (at) sunrise-tel (dot) (Bcom. (B (BThe previous OSX install package with CVS from end of last (Byear, which we've put up about 10 days ago, has been (Bdownloaded nearly 800 times by now. So, it seems there is (Bplenty of interest in Asterisk amongst Mac users. (B (B (B3) Other GUI tools still work in progress (B (BIvan Myvold and I have been working on a variety of OSX (BGUI tools for Asterisk as of late. Many people have been (Basking about the status of this already. However, other (Bthan the above mentioned Assistants this is still very (Bmuch work in progress. Nevertheless, here are some (Binsights for the curious ... (B (BIvan has built a very nice Asterisk console app which can (Bconnect to local or remote Asterisk servers and issue (Bconsole commands GUI style with returned data additionally (Bbeing displayed as a table view in a split window. The (Btable view allows it to do all the things OSX users are (Bused to do with tables, most notably reorder and sort by (Bany coloumn clicked on. We still need to teach it more (Bcommands though. (B (BConsole content will be fully searchable and there will be (Bfilters for debugging output and masking of any output (Bthat contains passwords. (B (B (BI have been working on an Asterisk configuration tool, (Bsome screenshots of which are at (Bhttp://www.sunrise-tel.com (for now) but I haven't been (Bable to get much further since there have been so many (Bother things popping up. I hope to be able to concentrate (Bback on this some time next week. (B (BAdri Vidal was kind enough to help us with the Icons, (Bsince we couldn't figure out how to get it from Photoshop (Bto ICNS format. Thanks Adri. (B (B (BLast but not least, if there are any Cocoa programmers or (Bany programmers who know how to use one of the various (BXYZ-to-ObjC bridges with their preferred language and who (Bwould like to contribute to some of the stuff we're doing, (Bplease contact me off-list. (B (B (BBefore I close, please allow me to ask for a little help (Bwith getting the OSX build of Asterisk listed at (BMacUpdate.com. I have submitted twice and sent email to (Bthem asking why it didn't show up in their catalog yet. I (Bgot a response from some Joel telling me they've never (Bseen anything. Sent them another email with URLs and all (Bthe rest of it, but still nothing and now silence. (BPerhaps, they don't like open source software but maybe if (Bthey get a few email messages of the kind "Why don't you (Blist Asterisk?" then they might change their mind ;-) (B (B (Brgds (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN$B!!(BJOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trouble compiling asterisk-addons MySQL
Yep.. I already have the headers and required files. Here is what I am getting now with my Make file also below it. [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include`ls *.c` cc -fPIC -I../asterisk -I/usr/local/mysql/include/mysql -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-linux/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 # # Asterisk -- A telephony toolkit for Linux. # # Makefile for CDR backends (dynamically loaded) # # Copyright (C) 1999, Mark Spencer # # Mark Spencer [EMAIL PROTECTED] # # This program is free software, distributed under the terms of # the GNU General Public License # MODS= CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I/usr/local/mysql/include/mysql CFLAGS+=-D_GNU_SOURCE INSTALL=install INSTALL_PREFIX= ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk MODULES_DIR=$(ASTLIBDIR)/modules # # MySQL stuff... Autoconf anyone?? # MODS+=$(shell if [ -d /usr/local/mysql/include/mysql ] || [ -d /usr/include/mysql ] || [ -d /usr /local/include/mysql ] || [ -d /opt/mysql/include ]; then echo cdr_addon_mysql.so; fi) CFLAGS+=$(shell if [ -d /usr/local/mysql/include/mysql ]; then echo -I/usr/local/mysql/include ; fi) CFLAGS+=$(shell if [ -d /usr/include/mysql ]; then echo -I/usr/include/mysql; fi) CFLAGS+=$(shell if [ -d /usr/local/include/mysql ]; then echo -I/usr/local/include/mysql; fi) CFLAGS+=$(shell if [ -d /opt/mysql/include/mysql ]; then echo -I/opt/mysql/include/mysql; fi) MLFLAGS= MLFLAGS+=$(shell if [ -d /usr/lib/mysql ]; then echo -L/usr/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /usr/local/mysql/lib ]; then echo -L/usr/local/mysql/lib; fi) MLFLAGS+=$(shell if [ -d /usr/local/lib/mysql ]; then echo -L/usr/local/lib/mysql; fi) MLFLAGS+=$(shell if [ -d /opt/mysql/lib/mysql ]; then echo -L/opt/mysql/lib/mysql; fi) all: depend $(MODS) install: all for x in $(MODS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done clean: rm -f *.so *.o .depend %.so : %.o $(CC) -shared -Xlinker -x -o $@ $ ifneq ($(wildcard .depend),) include .depend endif cdr_addon_mysql.so: cdr_addon_mysql.o $(CC) -shared -Xlinker -x -o $@ $ -lmysqlclient -lz $(MLFLAGS) depend: .depend .depend: ./mkdep $(CFLAGS) `ls *.c` -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Oleg A. Arkhangelsky Sent: Wednesday, July 28, 2004 9:48 AM To: [EMAIL PROTECTED]; James Freire Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Trouble compiling asterisk-addons MySQL Hello James, Wednesday, July 28, 2004, 5:23:16 PM, you wrote: JF [EMAIL PROTECTED]:/usr/src/asterisk-addons# make install JF ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` JF cdr_addon_mysql.c:33:19: mysql.h: No such file or directory JF cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory JF for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done You need to install libmysqlclient-devel (or alike) package with relevant header files. -- Best regards, Olegmailto:[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
RE: [Asterisk-Users] HELP! With Postresql
I installed Postresql and then recompiled Asterisk. I understood that Asterisk would see Postresql on the recompile and add it. Is there a way of checking? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Darragh Sent: Tuesday, July 27, 2004 10:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] HELP! With Postresql Have you actually compiled the pgsql CDR module in to Asterisk? On Wed, 2004-07-28 at 09:43, Martin Keding wrote: I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. Martin ___ 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 -- Stephen Darragh Technical Director Informed Technology Ph: +61 8 9380 4244 Fax: +61 8 9380 4354 ___ 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] MS SQL Free TDS
Help! I've been using mysql for cdr storage, I need to switch to MS SQL. I must be stupid or something but I cannot figure out how to setup the cdr_tds. I have FreeTDS configured properly, but my unixodbc is not working properly either... I'd be happy with either solution, but I'm in need of assistance. Luke Catranis ___ 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] error from asterisk
I am A few hours old with Asterisk. When I start up the Asterisk Server I get this error, *CLI Jul 27 00:58:18 WARNING[1150495040]: chan_oss.c:268 sound_thread: Read error on sound device: Resource temporarily unavailable Jul 27 00:58:24 WARNING[1150495040]: chan_oss.c:268 sound_thread: Read error on sound device: Resource temporarily unavailable Could anyone tekll me What this means ? Thanks Preeti Preeti Gopalan 404-526-6056 ___ 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 transfer bug in last CVS ?
After some more work : IAX phones cannot transfer outgoing channels when the dial string is like IAX2/ip They can when the dial string is like IAX2/recordname However it still fails with a recordname of 14 characters. ___ 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] Music On Hold - not working for me...
Hi all,
I'm trying to make some simple MOH (Music On Hold) working. So far I've failed
miserably - so I turn here for help.
Basically I've been using the wiki and all the sample confs I could from there
and via google.
The queue system seems to work fine with my limited setup. Just 2 IAX2 clients
where I keep Client B busy (by making it listen to mp3 via ext. 777) but logged
into the queue. Client A then calls the queue (tried both ext. 7320 and 6320)
and the announcements are fine (you are next in line etc.). When I make
Client B not busy - it starts ringing like it should on the queue. But I never
hear the MOH on Client A.
Also - calling 777 does play the mp3 fine - like it should - looped :)
Speaking of 777, I also did: chmod 755 /var/lib/asterisk/mohmp3/*
It's not really stopping me from rolling out this system - but it would be very
nice to have. Any help/pointers appriciated.
Thanks!
Various stuff that might be relevant...
zapata.conf
-SNIP-
musiconhold=default
-SNAP-
musiconhold.conf
-SNIP-
[classes]
default = mp3:/var/lib/asterisk/mohmp3
-SNAP-
extensions.conf
-SNIP-
[macro-queue1]
exten = s,1,Answer
exten = s,2,Queue(${ARG1})
[macro-queue]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,2
exten = s,4,ResponseTimeout,3
exten = s,5,Background(groovy)
exten = s,6,Queue(${ARG1})
[test]
exten = 6320,1,Macro(queue,Q320)
exten = 6330,1,Macro(queue,Q330)
exten = 6340,1,Macro(queue,Q340)
exten = 6350,1,Macro(queue,Q350)
exten = 6510,1,Macro(queue,Q510)
exten = 69000,1,Macro(queue,Q9000)
exten = 7320,1,Macro(queue1,Q320)
exten = 777,1,Answer
exten = 777,2,MP3Player(/var/lib/asterisk/mohmp3/trickme.mp3)
exten = 777,3,Goto(777,1)
-SNAP-
queues.conf
-SNIP-
[Q320]
announce-frequency = 5
announce-holdtime = yes
strategy = roundrobin
music = default
member = Agent/310,100
member = Agent/312,90
member = Agent/313,10
-SNAP-
outtake from full logfile at http://relay.dk/~avizion/asterisk/paste1.txt
PS: Should I attach this paste1.txt - or store it elsewhere?
--
avizion on irc.freenode.org #asterisk
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RE: [Asterisk-Users] Best Linux for Asterisk
I'm not religous about any particular flavor of Linux, but I am highly partial to Debian, for multiple reasons. As far as running *, I think one can simply to an apt-get install asterisk libpri zaptel and be ready to go. No, it won't be the absolute latest code, but the Debian community is pretty good about keeping packages updated. Debian, to me, is the easiest Linux/unix that I've ever been around. -Mark ___ 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] Best Linux for Asterisk
Hi Andy, Before Asterisk came into my life, I hadn't used Linux since RedHat 4.7. I did some research and decided to use one of the debian netinst images this go around, and I couldn't be happier. While it took me a day stumbling thru the packages and re-learning my way around, figuring out dependencies to get everything compiled and working etc... I've gotta say that the Asterisk + libpri + zaptel + tts stuff is rock solid, as is the system. I'm running Debian Woody with the 2.4 kernel. This system is also in a heavily used production environment within a software company. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Kirkland Sent: Wednesday, July 28, 2004 6:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux for Asterisk Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? Andy --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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] MGCP Caller ID
YES, because you could have an MGCP gateway device (more than one POTS line) ie. ours have 4 If so you would do something like this... [2084728800103]host = dynamiccontext = westcomllccallerid = "Jeremy Jones" 103nat = notransfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED]line = aaln/1 callerid = "Jeremy Jones #2" 104transfer = yescallwaiting = yesthreewaycalling = yescancallforward = yesmailbox = [EMAIL PROTECTED]line = aaln/2 ... etc... I do have a question for you though... I experimented with host=dynamic on the MGCP channel (we use MGCP here) I got it to work, but in this scenario, it was fatal, I'll explain and please tell me if you see the same thing. With host=dynamic, our MGCPend devicewould register with asterisk when powered up or when making the first call. All OK here, and asterisk would now remember (in memory) this registration, so any calls going back to the end device would be mapped appropriately. The fatality is that if asterisk is restarted, this "database of mapping" which was saved in memory; is now lost, so if a call came in and the end device was neverrebooted/restarted (to accomidate the asterisk restart) the mapping did not exist, as it was not saved in a "database" and the call would fail. So I switched back to host=ip.ip.ip.ip Do you see the same problem? Please let me know. Thanks Duane Cox - Original Message - From: Jeremy Jones To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:50 AM Subject: RE: [Asterisk-Users] MGCP Caller ID Hi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Wednesday, July 28, 2004 7:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP Caller ID Try this: mgcp.conf [2084728800103] host = dynamic context = westcomllc callerid = "Jeremy Jones" 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1Aha! Yup, that did the trick. So order matters there... Thanks,jeremy___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HELP! With Postresql
Subject: [Asterisk-Users] HELP! With Postresql I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. Martin Could you provide any details to your configuration and details on the errors that you see? It is a little hard to intuit from a blank page ;) dboyd(at)fullmoonsoft.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
Re: [Asterisk-Users] debian install zaptel
On Wed, 2004-07-28 at 06:25, Duane wrote: Jonathan Moore wrote: I saw the same problem on a customer install and because of short time frame we wiped system and moved over to Fedora, since we already has tested and used it in the past. I actually had the opposite experience where fedora wasted crap loads of my time and wiped the hdds and I had asterisk on debian up and running in minutes... While most all people know me as a Debianista, I have to say that in both this case and the one from debian to FC both have the same problem. Specifically, when you switched distros, you finally did what you where supposed to do to get the code working. The distro itself doesn't stop you from running asterisk, or really make it that much harder to install. -- 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] Best Linux for Asterisk
On Wed, 28 Jul 2004 10:22:53 -0400, Deon Rodden [EMAIL PROTECTED] wrote: It may sound bad, but I use Fedora Core 1. I don't think it sounds bad. I'm using FC2. I really like it. I haven't done a lot of the advanced stuff on it yet, but zaptel and Asterisk compiled fine with the stock kernel (after doing a yum update). I'm using ztdummy on a PII 350 and MeetMe conferencing works. Haven't installed Festival (I tried it once, and found it fairly unusable for an auto-attendent, and had no other uses in scripts for it). Will probably test MOH in the next couple of days. However, I installed using reiserfs (my preferred filesystem) and I installed all the updates and had to custom compile a new kernel (as the stock one that comes with Fedora is too screwy, and the sources aren't done right and certain programs wouldn't compile). However, once I did the updates and used a custom compiled kernel, everything runs fast/smooth on it. I haven't had an issue compiling anything Asterisk related, yet. I did have issues with my Redhat 9 server, and my OpenNA 1.0 server. Of course, I have put a lot of security and performance tweaks into my Fedora installation, had to make it less redhat but now it runs good. Obviously whatever works, works. However, this is what I did to get my system running on FC2: - Install FC2 *only* with [X] Development Tools, [X] Kernel Sources (not really needed as you compile against the build directory, but nice to have anyways), [X] Editors - Once installed, do a yum update - Reboot to make new kernel active - cd /usr/src/ - Verify that /usr/src/linux-26 is pointed to /lib/modules/`uname -r`/build/ ^^^ This should get updated when installing the new kernel, but always good to verify it did update the link and isn't pointing to the old build directory - Checkout Asterisk and Zaptel from CVS - cd /usr/src/zaptel ; make clean ; make linux26 ; make install - cd /usr/src/asterisk ; make clean ; make install I didn't have to do anything fancy to get this to work. Perhaps I just got lucky, but I think a lot of the problems people find on FC2 is not keeping the system clean at first. After doing this, I did yum install's for httpd, mysql-server and dhcpd. I even compiled phpmyadmin for mysql-server interface. Everything is still running great after at least 2 weeks of use. This is not a fancy computer either, scrap PII 350 parts with 256 MB of RAM (wait... 192MB, had a bad RAM stick in there...) Just a FYI. I think you'll find that as long as your favorite distro is decent, Asterisk will work. Agreed. The *best* Linux distro is the one you are most comfortable with. Thanks, Leif Madsen. http://www.asteriskdocs.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] MGCP Caller ID
Hi Duane (et alia), YES, because you could have an MGCP gateway device (more than one POTS line) ie. ours have 4 If so you would do something like this... [2084728800103] host = dynamic context = westcomllc callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1 callerid = Jeremy Jones #2 104 transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/2 ... etc... I have, actually, a gazillion 4-port mgcp devices from a (recently-obtained-by-8x8) company called Centile that I've _never_ been able to get to work properly w/* -- maybe this info'll help me here... ...snip... The fatality is that if asterisk is restarted, this database of mapping which was saved in memory; is now lost, so if a call came in and the end device was never rebooted/restarted (to accomidate the asterisk restart) the mapping did not exist, as it was not saved in a database and the call would fail. So I switched back to host=ip.ip.ip.ip Yeah, that's an issue here, too. We primarily have sip devices, though, at all our customer sites, so it's only a problem with _my_ phone internally, which so far doesn't bother me (I hate talking on the phone, anyway). If I just pick up the handset connected to the mgcp device hangup, that magic mapping is re-created. I'd love to be able to deploy some of these things, though, for our customers I really wouldn't like all the maintainance involved in setting up static dhcp assignments for all these mgcp devices tying addresses to each mgcp endpoint in mgcp.conf. We have, as I mentioned, tons of these mgcp thingies lying around waiting for use. Jeremy ___ 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] Desired Install in MotorHome
Paul Oster [EMAIL PROTECTED] wrote: I've got a client who [wants VoIP working over a very high-latency link]. So whats everyones opinion, worth exploring further, or am I wasting my time trying? Can you stick an Asterisk box at his end so you can speak IAX over the link? It may not help with the massive delays (which is going to be inherent in any kind of VoIP over the link) but the signalling should be a lot more reliable. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ 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] Best Linux for Asterisk
On Wed, 2004-07-28 at 08:13, Eric Kirkland wrote: Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? I'll try to avoid the RMS speech. At the core linux is just a kernel and a few glue apps. Most of the glue apps are all from GNU and deviat from one another by less than a percentage point. The only difference really introduced in a linux distro is how streamlined the install is, and a few of the admin tools. From recent comments about FC kernels, and my personal feelings about debian kernels, you are best off compiling a fresh stock kernel. OF course this advice is getting stale now as of the new kernel development model. -- 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: Re: [Asterisk-Users] Pickup zap channel already in use?
Hmmm I didn't notice any noise, but I was only focusing on connectivity. -Mark I'm getting a ton of noise on the channel just from the * side when I pickup my zap channel. Otherwise it works fine, if the other person in the house hangs up the noise goes away.. On Tue, 27 Jul 2004 23:30:08 -0400, Mark Woods [EMAIL PROTECTED] wrote: Kent wrote: On Tue, 27 Jul 2004 16:58:47 +, Mark Woods [EMAIL PROTECTED] wrote: I have to admit that your question interested me because I'm thinking of setting up the same thing. As of yet, though, I haven't found an answer to it. It's fairly simple when * has picked up, but I haven't really devoted much time to figuring out how to do it when it hasn't. So...let me work on it, and I'll let you know what I come up with. It's going to take at least a week, though, as I'm going to Oshkosh for the EAA Airventure on Thursday...but I'll see what I can come up with after that. -Mark A friend of mine who is another * user suggested using an extension with an empty Dial statement to connect my sip phone to the zap channel. I am going to try that tonight and see if that works. Let me know if you figure out anything else. Thanks! I actually got a chance to try it just now. Works like a champ! It has the added benefit of giving direct access, with dialtone, to the outside line, instead of having * dial. Here's what I put in my extensions.conf: exten = 4000,1,Dial(Zap/1/) exten = 4000,2,Congestion -Mark ___ 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
