[Asterisk-Users] Music on Hold
Having a weird issue with on hold music ... I do have mpg123 installed.
When requesting extension for testing, which is setup as:
exten = ,1,Answer ; Answer the line
exten = ,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
I recieve this err:
-- Executing MP3Player(SIP/100-26af, /sample-hold.mp3) in new stack
WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0
Not sure what's up...
Phillip
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Re: [Asterisk-Users] SS7 signaling/Softswitch
Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve Steve, I agree with you. If the discussion involves * then it should be here. In the case of your fax program I think some people who jumped in after the initial introduction might have thought it was totally separate and didn't make the connection. What I find really good about the fax discussion last week was that in the course of 48 hours it went from a non-working integration to functional in Asterisk. There is a tremendous resource base here... If we aren't interested in a discussion then the delete key or mail filters work wonders. Personally I read at least the beginning of all threads... Never know when a new idea or resource is mentioned. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:SIP Provider Question
Hi Alexander, Do you offer IAX wholesale termination or SIP only?? Can you send me your termination rates and charges?? Thanks.. NOC wrote: *David,* ** We has successfully tested last week interoperability between Asterisk and our SIP softswitch. We can definately help you with your project. Our company is New York based, so it will be very easy to get interconnection with you. We are mostly in wholesale business. Also you can get ,if needed, direct interconnection at our second POP in Zurich, Switzerland. Please contact me , if any interest. Alexander Kandelaki Stealth Telecommunications New York, NY ** *Mi**chael Bielicki [EMAIL PROTECTED] mailto:asterisk-users%40lists.digium.com /Wed, 1 Oct 2003 20:25:59 +0200/ * Previous message: [Asterisk-Users] SIP Provider Question http://lists.digium.com/pipermail/asterisk-users/2003-October/022358.html * Next message: [Asterisk-Users] grandstream phones and Transfer http://lists.digium.com/pipermail/asterisk-users/2003-October/022362.html * *Messages sorted by:* [ date ] http://lists.digium.com/pipermail/asterisk-users/2003-October/date.html#22359 [ thread ] http://lists.digium.com/pipermail/asterisk-users/2003-October/thread.html#22359 [ subject ] http://lists.digium.com/pipermail/asterisk-users/2003-October/subject.html#22359 [ author ] http://lists.digium.com/pipermail/asterisk-users/2003-October/author.html#22359 We can do that in the UK and in Poland and soon in more countries, but since you are US based I would recommend rather to talk to Jeremy at Nufone. On Wednesday 01 October 2003 8:30 pm, David Harris wrote: / Are there any sip providers out there providing full business telephone // service. Not just single line/residential service like I have seen with // vonage etc. // // For example take a company currently using a legacy pbx connected to the // PSTN with a PRI. I would like to replace this setup with a data T1, an // asterisk box, and some SIP Phones then pass all calls (local and long // distance) directly asterisk box to the SIP provider. Also I would need // to able to port the companies existing numbers over to the SIP provider // in order receive incoming calls. // // Thanks, // David / -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
Jan Janak wrote: I experimented a little bit and Asterisk behind NAT with SIP works. I created an account at iptel.org and use that account for outbound SIP traffic from Asterisk. Great! I copied your information for other users to the Wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind console keyboard, but anyway... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confuson on iax calls (register or not?)
Rich Adamson wrote:
Think I'm a little confused on registering an iax connection; could
someone enlighten me?
I guess the real question is... when two * machines are going to rely
on an iax link (each with their own dial plan), do both machines have
to register with each other (eg, both need a 'register' statement)?
Or, will a single machine doing the registering cause the opposite
machine to recognize the registration, and allow calls to be originated
in both directions?
If so, assume machine B registers with machine A (machine B has the
register statement). Then, in machine A's extensions.conf dial plan,
what would the statement similar to
exten = _6X.,1,Dial(IAX/npi-off:[EMAIL PROTECTED]/${EXTEN-1})
look like? (I'm assuming the above might use the context for which
machine B registered with?)
AFAIK a register directive is only required when one of the servers has
a dynamic IP address which cannot be resolved from the outside or is
behind NAT..
So if both your servers are on the open internet then you probably don't
need to have any register directives set..
Later..
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[Asterisk-Users] ReplayTV connecting through Asterisk box
Has anyone had any luck getting a ReplayTV DVR box to connect
through an Asterisk box? Mine seems to dial just fine, but can't
negotiate a connection. I am using:
exten = _95380024,1,Dial(Zap/1/${EXTEN:1},120,d)
exten = _95380024,2,Congestion
I don't have any problems doing a fax though my system.
For this setup, I am running a simple Digium developer's kit
on a P3-700 256MB. I have a DG104S and ATA-186 too, but I'm
not using them for this. The ReplayTV box is connected directy
to the TDM400P. I am running a relatively old build:
Asterisk CVS-07/10/03-23:27:00
Any thoughts would be appricated.
-- John
(As a humorous aside, I just spent a few hours tonight trying to figure out
why I could not do an outside call to a particular phone number. It turned
out that the '2' button on the phone I was using is not working quite right.
A different phone worked fine.)
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Re: [Asterisk-Users] SS7 signaling/Softswitch
OK. If are talking about masses of messy details, then I agree with you. Regards, Steve CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confuson on iax calls (register or not?)
Hi! Think I'm a little confused on registering an iax connection; could someone enlighten me? Not so long ago I had the same question - I guess a slight change of explanation in iax.conf might help to prevent this. I guess the real question is... when two * machines are going to rely on an iax link (each with their own dial plan), do both machines have to register with each other (eg, both need a 'register' statement)? AFAIK a register directive is only required when one of the servers has a dynamic IP address which cannot be resolved from the outside or is behind NAT.. Indeed. A couple of more related hints (as you can see I have been through this): - only type=peer can register at the other side (maybe also friends) - registration only (!) works if that peer has been set to host=dynamic - use iax show registry to learn about my (the servers') own registration (status column: see qualify below) - while you can DIAL using the username (in place of the hostname) of the remote peer that registered at your server you cannot register vice-versa using his username, even if he already registered with you (i.e. you won't be able to tie two * together that way) - take a look at qualify=yes to have the fixed-IP * run regular presence checks on the dynamic *. Note that it might (?) be wise to NOT use qualify because if the host appears (!) to be down/too slowly connected then * will refuse to direct calls to it via IAX. I still have an unsolved issue where I get UNKOWN as status with qualify=yes ALTHOUGH the other server is actually up and alive (and I even have a standing SSH working nicely). However, maybe this is linked to the other reports on the list about the current CVS breaking IAX calls...? - in iax.conf you enter only one port, i.e. you need to choose between IAX and IAX2. Still registration will try to register using both ports - still I repeatedly see cases where only 5036 worked and IAX2 didn't (same server,s same setup). Greetings, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Man, getting the masses of messy details out in the open is what this is all about. Let it flow and we all get better and can do better! Steve Underwood wrote: OK. If are talking about masses of messy details, then I agree with you. Regards, Steve CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confuson on iax calls (register or not?)
Think I'm a little confused on registering an iax connection; could
someone enlighten me?
I guess the real question is... when two * machines are going to rely
on an iax link (each with their own dial plan), do both machines have
to register with each other (eg, both need a 'register' statement)?
Or, will a single machine doing the registering cause the opposite
machine to recognize the registration, and allow calls to be originated
in both directions?
If so, assume machine B registers with machine A (machine B has the
register statement). Then, in machine A's extensions.conf dial plan,
what would the statement similar to
exten = _6X.,1,Dial(IAX/npi-off:[EMAIL PROTECTED]/${EXTEN-1})
look like? (I'm assuming the above might use the context for which
machine B registered with?)
AFAIK a register directive is only required when one of the servers has
a dynamic IP address which cannot be resolved from the outside or is
behind NAT..
So if both your servers are on the open internet then you probably don't
need to have any register directives set..
Okay, let me try a specific example (maybe a poor one) that might help me
understand.
Server A has a static registered address and is considered a production
office machine. It has pstn connections, VM, iaxtel, addressable via
dns, etc. (Changes to the configuration of this machine can only be made
once per month.)
Server B is a laptop running * that can be carried from one client's
location to another for demo purposes. Set the demo machine in the
client's conference room, connect to the Internet (assume no NAT),
configure a couple of 7960's (as an example) on the conference room
table next to the laptop. Assume the two 7960's always use ext 3500 and
3501. The registered address at a client's location is obviously not the
same from one client to another.
Now for demo purposes,
1. Server B registers with Server A via iax. That's easy to understand
regardless whether nat is involved or not, and making calls from the
sip phones at the client's location to phones registered with
server A (office) is clear. An entry such as:
exten = _6X.,1,Dial(IAX/office:[EMAIL PROTECTED]/${EXTEN-1})
in extensions.conf will work nicely since the server A is addressable
and remains static at all times.
2. Server A (without making any * config changes specific to the laptop
addressing at the client's location) wants to place a call from a
sip phone registered with server A (office), to one of the 7960's
(ext 3500) registered with server B (laptop).
What would the entry look like on Server A (office) that would allow calls
to be sent to Server B (laptop) when the specific IP address (and/or DNS
name) of the laptop is unknown? (Or, rephrasing the question, can this
example call be forwarded to Server B using a reference to a iax context
(that is either active or not active based on Server B registration) rather
then using an exten =... statement that specifies an exact IP address?)
Although the above example might be rather poor, the same config issue
can happen with off-site disaster recovery configs, etc.
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[Asterisk-Users] Asterisk Modules
Hello. Can somebody give me a link to an very good documentation about writing Aterisk modules ? And another question. I'm new to asterisk and I want to find out if there is posible to write my own function to use in contexts.. Like goto , background , play , etc.. If it is , how can I do that ? Regards Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail.conf in MySQL is not functioning
Hi, Here is the error from Asterisk messages log file that I forgot to put in. Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: Oct 25 10:55:11 WARNING[19474]: File app_voicemail2.c, Line 2388 (vm_execmain): Couldn't read username CLI debug output is as follows: Executing VoiceMailMain2(SIP/2205-3df0, ) in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = any) -- Playing 'vm-incorrect' -- Playing 'vm-password' -- Incorrect password '2421' for user '2205' (context = any) -- Playing 'vm-incorrect' Here are my configs In extensions.conf I am using Voicemail2 and VoiceMailMain2 that has support for MySQL exten = 8500,1,VoiceMailMain2 In voicemail.conf I have the MySQL connectivity settings in [general] dbhost=localhost dbname=asterisk dbuser=someuser dbpass=somepass As well in voicemail.conf I have commented out the entire [default] section, and mailboxes. I do have MySQL working with CDR MySQL from asterisk-addons thanks, John Haigh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower CPU linux system, Asterisk runs at 0.1% - both without any active channels... Any ideas, anyone recognizing the problem? Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) FWIW, I'm running yesterday's cvs on two RH9 systems just fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monitoring the asterisk and safe restart
I use daemontools ( http://cr.yp.to/daemontools.html ). Here, for instance, is my '/service/asterisk/run' script. You only need the first uncommented line and the last uncommented line. You can ignore all of the networking stuff, but I wanted to know, if anyone else happened to see this, if someone knew if zaptel's PPP device can use authenticated PPP (PAP/CHAT) in this case. I had to get a special setup from my ISP to get a frame without PAP authentication. /service/asterisk/run: #!/bin/bash # # Script to unload and reload asterisk and related networking functions # # Direct stderr to stdout so daemontools logger puts it all in log/main exec 21 # Bring down network connections so ztcfg can run ifconfig pvc0 down ifconfig hdlc0 down ifconfig hdlc1 down ifconfig hdlc2 down /etc/init.d/monmotha stop /etc/init.d/monmotha zap # Remove modules rmmod wcfxs wcfxo tor2 zaptel # Reload modules modprobe tor2 modprobe wcfxo modprobe wcfxs sleep 3 # wcfxo doesn't like things too fast, so wait 3 sec # Configure WAN cards ztcfg - sleep 3 # wcfxo doesn't like things too fast, so wait 3 sec # ## Configure Frame to ISP sethdlc hdlc0 mode fr-ansi create 16 ifconfig hdlc0 up ifconfig pvc0 local public ip pointopoint gateway ip up route add default gw gateway ip # ## Configure WAN link to remote office 1 sethdlc hdlc1 mode hdlc ifconfig hdlc1 192.168.1.1 pointopoint 192.168.1.2 up route add -net 10.90.45.0 netmask 255.255.255.0 gw 192.168.1.2 # ## Configure WAN link to remote office 2 sethdlc hdlc2 mode ppp ifconfig hdlc2 192.168.2.1 pointopoint 192.168.2.2 up route add -net 10.90.33.0 netmask 255.255.255.0 gw 192.168.2.2 # ## Configure route to the rest of internal network route add -net 10.90.0.0 netmask 255.255.0.0 gw 10.90.31.1 metric 2 # ## Start Firewall /etc/init.d/monmotha start # ## Start Asterisk exec asterisk - On Friday, 03 October, 2003 19:43, [EMAIL PROTECTED] wrote: Hi List, I am sorry that I may bring the old question to the community. My question is 1. How can we determine if asterisk is working normally or not ? what kind watchdog process do we have at this moment ? 2. In case the running asterisk is mulfucntion, is there any available way to auto restart asterisk ?? Please advise if you could. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP auth
Hello.. There is another way of doing SIP auth other then manually add the user passwords to sip.conf ? I'm talking about radius or postgres.. Regards Alex
[Asterisk-Users] ATA-186 Troubels
Hello all, Things are going well. I've even unlocked that extra ATA-186 I had lying around - however, a problem I've noted. When queing on-hold music using dedicated extension 211, I get the following error: NOTICE[1225991360]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Is it possible to change or disable this in the ATA? I haven't found anything pertaining to this error on the Asterisk boards. Regards, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Sometimes, if * dies, mpg123 keeps running and eats all memory. Try to stop *, kill all mpg123 instances and try again. Also, you can modify your start script to kill all mpg123 instances before * starts 'killall -9 mpg123' Regards, Gus - Original Message - From: TeleSIP [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 6:41 PM Subject: Re: [Asterisk-Users] Asterisk on FreeBSD - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 3:11 PM Subject: Re: [Asterisk-Users] Asterisk on FreeBSD My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower CPU linux system, Asterisk runs at 0.1% - both without any active channels... Any ideas, anyone recognizing the problem? Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) Same here with mpg123. Once time we saw 2 extra mpg123 processes eating 99% of the CPU. No idea why they were there. FWIW, I'm running yesterday's cvs on two RH9 systems just fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
--- Olle E. Johansson [EMAIL PROTECTED] wrote: My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower CPU linux system, Asterisk runs at 0.1% - both without any active channels... Any ideas, anyone recognizing the problem? You mean both are in the idle state, no active calls? That should be a big clue. My guess is that the difference you see is due to different implementations of pthreads. Have you tried running a debugger? Most of us don't have BSD systems and can't help. But if it were me a debugger would be my first step. IMO gdb is a pain to use. I like ddd. See here: www.gnu.org/software/ddd/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP auth
Alexandru Coseru wrote: Hello.. There is another way of doing SIP auth other then manually add the user passwords to sip.conf ? There are scripts that generate the text files and a patch that adds the functionality to hide passwords and replace them with MD5 digests, but as far as I know, there's no way to get Asterisk to authenticate by questioning a database. I'm talking about radius or postgres.. Radius was mentioned on the list recently. Keep listening to the list for more information on this subject :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extensions Problem
Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten = 1637,1,Dial(SIP/100) Exten = _NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _NX,2,Congestion Exten = _1NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _1NX,2,Congestion These extensions allow us to utilize our SIP provider - ONLY when being dialed from a regular telephone attached to a Cisco ATA-186. Our Cisco 7960 only allows us to dial 4 charachters before it tries dialing. So, I assume we need to implement 9, and the number. However, when I do this, the 9 gets passed on to our SIP provider, which tries to dial 9NXX, and all goes to hell. Question - is there a way to allow 9 in the dialing plan, without having it be passed to the sip provider. Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
- Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 3:11 PM Subject: Re: [Asterisk-Users] Asterisk on FreeBSD My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower CPU linux system, Asterisk runs at 0.1% - both without any active channels... Any ideas, anyone recognizing the problem? Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) Same here with mpg123. Once time we saw 2 extra mpg123 processes eating 99% of the CPU. No idea why they were there. FWIW, I'm running yesterday's cvs on two RH9 systems just fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ReplayTV connecting through Asterisk box
Has anyone had any luck getting a ReplayTV DVR box to connect through an Asterisk box? Mine seems to dial just fine, but can't negotiate a connection. I am using: I would suggest NOT using the agressive echo cancellor. I think it buggers up modems in a big way. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower CPU linux system, Asterisk runs at 0.1% - both without any active channels... Any ideas, anyone recognizing the problem? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lucent Partner extension to X100P
Did you try fxsls or gs? On Sun, 26 Oct 2003, Jean-Philippe Lord wrote: Hi All... I'm currently trying to have an extension of my Lucent Partner phone system connected to Asterisk using an X100P. The issue I'm having is that the Lucent Partner analog port connection have different ring and dialtone than the one specified for US in indications.conf. Anyone have accomplished this connection before. I need to know what I should use for the different setting in indications.conf. Otherwise, I was able to record the ringing pattern using ztmonitor. How do I analyze this sound file to find out the required information. Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] important feature missing?!
one of the common features in PBX devices is the ability to set a time limit on duration of calls (esp. outgoing, for each station) and usually a warning beep is played few seconds before time runs out. as far as i could understand it's possible to set a time limit on calls using something like: exten = 2000,1,AbsoluteTimeout(20) but that's only on *incoming* calls and even so, the call is disconnected without any warning beeps or anything. so i was wondering, am i missing something or is this feature really not availabe in asterisk? [moved to asterisk-users, where this belongs] You are correct about there being no whisper functionality or timers to play a beep or audio file at a predetermined interval. You are incorrect about the usage of AbsoluteTimeout. It can be used anywhere to limit the length of a call leg. Additionally, the T extension can be used to play back a message after the AbsoluteTimeout is reached, though I suspect that you cannot re-connect the two call legs back together again. Search the archives for details. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A software FAX modem
This sounds like the fax resolution is incorrect. Basically, there are only two resolutions for faxes, normal and fine. The only difference in these two is the number of lines, or the Y dimension. With fine resolution, you simply have twice the lines. Unfortunately, I do not believe there is any header information telling which resolution the file is. The resolution _is_ communicated before sending the fax, however, as part of the initial communication negotiation. This basically means that, if it does not yet have the facility, the softfax application needs to record what resolution the fax is. On Wednesday, 22 October, 2003 10:49, Steven Critchfield wrote: Figured the group would like to hear this. I just faxed a sample document from a real fax machine to asterisk semi successfully. I'll consider it just semi successfully for now because either I haven't found a viewer that puts the image in proper aspect ratio or the storage is screwy. I'm thinking it may be the fact that image apps expect the file to be in X by X dpi not X by Y. Otherwise it was readable. Also I was able to take the resulting tiff file and create a sample call file that then sent the file back out to the real fax machine successfully. The output was nearly identical to the original with the exception of being darker. I'll attribute that to cheap fax machine with crappy scan head. Otherwise, Great job. So far this is my bug list. 1. Makefile uses a include and library directory from /home/steveu. 2. Shouldn't make install for the spandsp library put the headers and libraries in the proper locations so we don't have to make special include links? Basically if #2 is fixed, then #1 will not need those paths. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr_mysql Voicemail or VoiceMail2?
Can I use Asterisk with MySQL Voicemail or do I need VoiceMail2? Uriel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
Jojojo! Funny guy... - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 1:57 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch press delete msg or delete thread and move on On Sun, Oct 26, 2003 at 01:34:04PM -0300, CW_ASN wrote: You see, guys? Some people loves to see masses of messy details and other people don't... What can we do? Regards, Gus - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 6:58 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Man, getting the masses of messy details out in the open is what this is all about. Let it flow and we all get better and can do better! Steve Underwood wrote: OK. If are talking about masses of messy details, then I agree with you. Regards, Steve CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help: mixing monitored in out files
Hi, There is an email in the mail archives from asterisk-users-admin with a script to mix two monitored files. I was trying that script but the wmix always says : Unable to open display "(null)". I am not using X, which wmix requires. Is there any other tool or any help on making a single file out of the in and out monitored files ?? TIAAzher Do you Yahoo!? Exclusive Video Premiere - Britney Spears
Re: [Asterisk-Users] Music on Hold
MP3Player is not the way to have Music on Hold... Please do a test in this
way:
exten = 2091,1,Answer
exten = 2091,2,Wait,1
exten = 2091,3,MusicOnHold,default
And the musiconhold.conf:
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
;loud = mp3:/var/lib/asterisk/mohmp3
;random = quietmp3:/var/lib/asterisk/mohmp3,-z
Hope this helps.
Gus
- Original Message -
From: Phillip Jackson, Director of IT [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 26, 2003 4:14 AM
Subject: [Asterisk-Users] Music on Hold
Having a weird issue with on hold music ... I do have mpg123 installed.
When requesting extension for testing, which is setup as:
exten = ,1,Answer ; Answer the line
exten = ,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
I recieve this err:
-- Executing MP3Player(SIP/100-26af, /sample-hold.mp3) in new stack
WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0
Not sure what's up...
Phillip
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Re: [Asterisk-Users] NuFone International Calls
Do you know why there are two different possible contexts? Of course it would seem a little strange to put somebody outside the US into the NANPA context rather than the WORLD one ... Michael On Mon, 27 Oct 2003 [EMAIL PROTECTED] wrote: TOP POSTING MADNESS continues... you need to be part of the WORLD context, and not just NANPA, otherwise 011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify -wasim On Sun, 26 Oct 2003, Michael T Farnworth wrote: Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. and I still trimmed the bush... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
Don't forget the equally important host stamp on the file. That allows you to write two different files at precisely the same time on a shared filesystem (e.g., NFS) with no race conditions. On Tuesday, 21 October, 2003 13:37, Andrew Kohlsmith wrote: There is a C Library function that will return a unique file name. (see man mkstemp) That's the best way to go. It is generally a bad design to encode any information in a file name. Better to simply use the file's date/time stamp to order the messages. I was speaking with tclark on IRC about this this past weekend. What is wrong with using Maildir/ type interfaces for voicemail? Maildir is a very straightforward, scalable and distributable way of storing things like email (and voicemail). Each mailbox has this format: ./ tmp/ cur/ new/ When a new voicemail is created, you mkstemp in tmp/ and create the file. Once it's done, you mv it to /new. When it's listened to or otherwise accessed, it's mv'd to cur where it stays until deletion. So to recap: create and manipulate in tmp/, move to new/ once done. When no longer new, move to cur/ and leave there. No funky locking, totally NFS safe and very fast, since each voicemail is just a file. There's no patents or any kind of software encumberances to this technique, either. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions Problem
quote who=Phillip Jackson
So, I assume we need to implement 9, and the number. However, when I
do this, the 9 gets passed on to our SIP provider, which tries to dial
9NXX, and all goes to hell.
Question - is there a way to allow 9 in the dialing plan, without having
it be passed to the sip provider.
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
--
END OF LINE
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Re: [Asterisk-Users] Asterisk Modules
I think the best start point is the main documentation of *, do a 'make progdocs' in your source directory. Regards, Gus - Original Message - From: Alexandru Coseru [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 10:34 AM Subject: [Asterisk-Users] Asterisk Modules Hello. Can somebody give me a link to an very good documentation about writing Aterisk modules ? And another question. I'm new to asterisk and I want to find out if there is posible to write my own function to use in contexts.. Like goto , background , play , etc.. If it is , how can I do that ? Regards Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch/ Unofficial Forums
Patrick, We have started unofficial Asterisk Forums : http://asterisk.xvoip.com I think such web-presence will help all parties to participate and exchange information on all cases, starting from SS7 compatibility, ending with business solutions, proposals, RFQ, etc. I had this morning quick chat with Asterisk Mark about making official face for the forums, so far I din;t finished my conversation with him, hopefully we will finalize everything on Monday. In all cases, unofficial web-based forums for Asterisk-related questions (anykind of questions : technical, business, general) is opened. I welcome everybody to join this forum and let's get moving this Asterisk project forward , by exchanging ideas, infos about it. Also we are looking for volunteers to handle and moderate some topics on Forum. Please contact if you are interested : [EMAIL PROTECTED] URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register - Original Message - From: Patrick [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 1:12 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Sun, 2003-10-26 at 06:40, CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus How about starting another list: asterisk-ss7 I would also like to see an asterisk-business list where * based solutions providers and telco's and their wannabee versions can find each other to discuss e.g. interconnection, termination and other business opportunities. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extensions Problem
Phillip, exten = _9NX,1,StripMSD,1 Exten = _NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _NX,2,Congestion Should work Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Phillip Jackson Sent: 26 October 2003 23:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Extensions Problem Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten = 1637,1,Dial(SIP/100) Exten = _NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _NX,2,Congestion Exten = _1NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _1NX,2,Congestion These extensions allow us to utilize our SIP provider - ONLY when being dialed from a regular telephone attached to a Cisco ATA-186. Our Cisco 7960 only allows us to dial 4 charachters before it tries dialing. So, I assume we need to implement 9, and the number. However, when I do this, the 9 gets passed on to our SIP provider, which tries to dial 9NXX, and all goes to hell. Question - is there a way to allow 9 in the dialing plan, without having it be passed to the sip provider. Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lucent Partner extension to X100P
Tried both and still won't work... It seems that the X100P won't detect the call coming in due to the unusual ring cadence. Outside call thru the Zap channel works as it gets the regular dialtone from the phone line. JP On Sunday, October 26, 2003, at 04:43 PM, Brian West wrote: Did you try fxsls or gs? On Sun, 26 Oct 2003, Jean-Philippe Lord wrote: Hi All... I'm currently trying to have an extension of my Lucent Partner phone system connected to Asterisk using an X100P. The issue I'm having is that the Lucent Partner analog port connection have different ring and dialtone than the one specified for US in indications.conf. Anyone have accomplished this connection before. I need to know what I should use for the different setting in indications.conf. Otherwise, I was able to record the ringing pattern using ztmonitor. How do I analyze this sound file to find out the required information. Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
I experimented a little bit and Asterisk behind NAT with SIP works. I created an account at iptel.org and use that account for outbound SIP traffic from Asterisk. Great! I copied your information for other users to the Wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind console keyboard, but anyway... There has been a fair amount of discussion on the list as to whether nat works with various/different configurations of sip phones with *. The exact configuration required is highly dependent on a number of technical factors that must be well understood before anyone can make a generic statement relative to whether it works or doesn't work. Without that understanding, practically every statement made on the list has been based on opinion and/or some trial error methodology that has resulted in a working example. (Nothing wrong with that, but the majority of the postings leave out critical info that causes the next person to attempt the same implementation but fails, and additional questions are generated.) The critical information needed to understand nat config's include: 1. Is * behind a nat box, sip phone behind a nat box, or both? 2. Is the nat box sip aware? 3. Can the nat box be programmed to forward a static range of ports to the inside? 4. Are there two nat boxes involved (one at each end of an expected sip-based connection)? 5. Does the sip phone support nat (eg, play nice with headers)? 6. Does * support nat (eg, play nice with headers) and is it config'ed? 7. Are there timers involved at either end of a nat traversal that are intended to keep nat table entries from timing out? 8. If so, what are the actual timeout values used for the specific nat box, and are sip end-point timers less then those of the nat box? (Don't assume all sip phones with nat functions are equal.) 9. What is the nat impact of a sip phone that has been configured to re-register every 60 seconds? 10. What is the range of rtp ports expected by the sip phone (eg, 7960's range from 16384 to 32766, but can be changed; xten uses 8000 to 8012 or something like that)? 11. Can the user implement iax (instead of sip) between end points? 12. When nat is found to function correctly, which end originated the nat traversal (makes a BIG difference)? And, probably another half dozen technical parameters that I'm forgetting to mention. I've spent many years working with corporate clients in more then 40 states diagnosing networking issues, doing protocol analysis, etc, and have seen a large number of nat boxes. The nat implementations from various vendors range from very basic translation tables to some rather sophisticated functions. And, just because a nat implementation comes from a well-known vendor doesn't mean anything (even Cisco has problems with no nat timeouts in certain boxes today). With that said, here's a couple of high-level examples that could work but these are not based on actual lab tests, etc. 1. If * is behind a nat box and * inititiates a tcp/udp conversation with a non-nat'ed address, some form of timer-based keep alive packet will keep the nat-box-table-entries active allowing the implementation to work. (Obviously assumes equipment can support sip header functions.) What are some of the configuration issues that may need to be addressed? a. limit the port numbers that can be used by * (rtp.conf) b. limit the port numbers that can be used by the sip phone. c. may still need to map the specific rtp port range in the nat box depending upon the nat box functionality. d. probably define nat=yes within *. (The real issue here is which end initiated the conversation and what is used to keep the nat translations active. I think we've already heard some folks doing this with certain Internet-based companies, but the postings left out a bunch of technical configuration data on both ends.) 2. * = nat = Internet = nat = sip phone Implement a combination of #1, above, at both ends assuming the end-point equipment has the capability to be configured (including the sip phone, nat boxes, etc). What tends to aggravate nat implementations are those NAT boxes that also implement PAT (port address translation), and the box vendor doesn't bother to hint at it in their documentation. (There are a very large number of networking folks that don't understand this, and its probably safe to assume 99.99% of the user community has never heard of it.) The PAT issues usually end up with someone suggesting sip phone #1 works but #2 doesn't and they are configured exactly the same. Or, call #1 works but call #2 fails. (And then the next person on the list says it works fine for them, but doesn't mention who's nat box he's using or what it's actually doing from a technical perspective.) I'd bet a small amount of money that
Re: [Asterisk-Users] SS7 signaling/Softswitch
press delete msg or delete thread and move on On Sun, Oct 26, 2003 at 01:34:04PM -0300, CW_ASN wrote: You see, guys? Some people loves to see masses of messy details and other people don't... What can we do? Regards, Gus - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 6:58 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Man, getting the masses of messy details out in the open is what this is all about. Let it flow and we all get better and can do better! Steve Underwood wrote: OK. If are talking about masses of messy details, then I agree with you. Regards, Steve CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NuFone International Calls
Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. I am getting: -- Executing Dial(SIP/phone1-adc5, IAX2/[EMAIL PROTECTED]/011441942XX) in new stack -- Called [EMAIL PROTECTED]/011441942XX WARNING[131081]: File chan_iax2.c, Line 4160 (socket_read): Call rejected by 65.127.126.42: No such context/extension -- Hungup 'IAX2[NuFone]/2' == No one is available to answer at this time -- Executing Congestion(SIP/phone1-adc5, ) in new stack I have tried dropping the 011 and jumping straight to the country code but that doesn't work either. Does anybody have any suggestions? Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Troubels
Yes its called CN. www.bkw.org/~brian/cisco/ata.html check audiomode and connectmode RFC 3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) bkw On Sun, 26 Oct 2003, Phillip Jackson, Director of IT wrote: Hello all, Things are going well. I've even unlocked that extra ATA-186 I had lying around - however, a problem I've noted. When queing on-hold music using dedicated extension 211, I get the following error: NOTICE[1225991360]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Is it possible to change or disable this in the ATA? I haven't found anything pertaining to this error on the Asterisk boards. Regards, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help: mixing monitored in out files
Hi, There is an email in the mail archives from asterisk-users-admin with a script to mix two monitored files. I was trying that script but the wmix always says : Unable to open display (null). I am not using X, which wmix requires. Is there any other tool or any help on making a single file out of the in and out monitored files ?? TIA Azher soxmix will probably work for you. It comes as a standard part of the most recent versions of sox. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
Rich Adamson wrote: I experimented a little bit and Asterisk behind NAT with SIP works. I created an account at iptel.org and use that account for outbound SIP traffic from Asterisk. Great! I copied your information for other users to the Wiki. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind console keyboard, but anyway... There has been a fair amount of discussion on the list as to whether nat works with various/different configurations of sip phones with *. The exact configuration required is highly dependent on a number of technical factors that must be well understood before anyone can make a generic statement relative to whether it works or doesn't work. Without that understanding, practically every statement made on the list has been based on opinion and/or some trial error methodology that has resulted in a working example. (Nothing wrong with that, but the majority of the postings leave out critical info that causes the next person to attempt the same implementation but fails, and additional questions are generated.) Rich, Thank you for your additional information on the NAT/VoIP issue. Is it ok with you if I add it to the Wiki? As you say, we need to collect information and compose a data base of what works and what's not working in certain circumstances. Jan got * - SER working, I can't. We have different NAT:s. To try to solve my problem I made sure his solution was documented so far. There's no silver bullet here. With NATs, we've built a network without end-to-end connectivity and we need to patch it up to get VoIP working on an IPv4 network with NATs in every corner. I just hope that IPv6 will make life easier for the next generation of VoIP users. Right now, we need to understand all variables. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) FWIW, I'm running yesterday's cvs on two RH9 systems just fine. i had a problem with asterisk consuming all the resources available on redhat 9. it would occur roughly every 24 hours or so - and would cause all sorts of problems. when a new channel opened up it fought for resources for a few seconds - so no speech could be heard, then when it could grab enough resources to process the channel it would... but the quality would be terrible. it can be solved with this: export LD_ASSUME_KERNEL=2.4.1 so now thats in my asterisk init script before actually starting asterisk. since doing this i havent had a problem (3 weeks ago). duncna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA-186 Troubels
Brian, Awesome. Thanks - this voip stuff is too cool. Regards, Phillip -- Phillip Jackson - [EMAIL PROTECTED] President, The Jackson Group - Intelligent IT. (TM) Ph 410.320.2138 Fx 443.321.8713 Returning violence for violence multiplies violence, adding deeper darkness to a night already devoid of stars. Darkness cannot drive out darkness; only light can do that. Hate cannot drive out hate. Only love can do that. - MLK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian West Sent: Sunday, October 26, 2003 3:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ATA-186 Troubels Yes its called CN. www.bkw.org/~brian/cisco/ata.html check audiomode and connectmode RFC 3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) bkw On Sun, 26 Oct 2003, Phillip Jackson, Director of IT wrote: Hello all, Things are going well. I've even unlocked that extra ATA-186 I had lying around - however, a problem I've noted. When queing on-hold music using dedicated extension 211, I get the following error: NOTICE[1225991360]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Is it possible to change or disable this in the ATA? I haven't found anything pertaining to this error on the Asterisk boards. Regards, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
I had similar problems, and were related to dtmfmode=inband in sip.conf - Original Message - From: duncan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 9:15 PM Subject: Re: [Asterisk-Users] Asterisk on FreeBSD Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) FWIW, I'm running yesterday's cvs on two RH9 systems just fine. i had a problem with asterisk consuming all the resources available on redhat 9. it would occur roughly every 24 hours or so - and would cause all sorts of problems. when a new channel opened up it fought for resources for a few seconds - so no speech could be heard, then when it could grab enough resources to process the channel it would... but the quality would be terrible. it can be solved with this: export LD_ASSUME_KERNEL=2.4.1 so now thats in my asterisk init script before actually starting asterisk. since doing this i havent had a problem (3 weeks ago). duncna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] alpha/numeric paging
I am looking for help in writing the scripts for numeric paging via Asterisk. Here's the general flow: answer lookup D-I-D if on prompt with please enter your telephone number store in variable prompt please hangup now Shell to SNPP executable and send message to server. I have the code in C for the command line SNPP client. Three parameters are passed: server ID message. James Taylor MetroTel 903-793-1953/1956 Sent via the KillerWebMail system at mail.metrotel.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone International Calls
On Mon, 27 Oct 2003 [EMAIL PROTECTED] wrote: On Sun, 26 Oct 2003, Michael T Farnworth wrote: Do you know why there are two different possible contexts? Of course it would seem a little strange to put somebody outside the US into the NANPA context rather than the WORLD one ... its not where you ARE, its where you're calling... NANPA gives you access to the contigous-US-48, whilst the WORLD pretty much leaves you at your own peril... (ofcourse, none of this is official NuFone, best let jerjer advise you accordingly) I really meant it is a little strange to put somebody who lives outside the US in a context which means they can only call US numbers, because they will almost certainly need to make calls to non-US numbers. I do want to make some US calls, but I did also request, and receive, a copy of the rates for all countries. If nothing else that was a very strong indicator that international calls were on my mind. Even the NANPA context leaves me at my own peril, but I guess it would take somebody longer to use up my money. Of course I would be able to watch out for this if the Subscriber Management System link didn't give me the message 'nothing to see here'. I don't suppose you know where account details are obtained from? By the way, what is jerjer, some sort of play on the name Jeremy McNamara, or his email address jj? Thanks, Mtf - wasim -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone International Calls
TOP POSTING MADNESS continues... you need to be part of the WORLD context, and not just NANPA, otherwise 011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify -wasim On Sun, 26 Oct 2003, Michael T Farnworth wrote: Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. and I still trimmed the bush... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
On Sat, 2003-10-25 at 18:49, Ken Godee wrote: You did do a make clean first before recompiling? Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk directories and re-checking them out. Then I decided it might be a heat issue, so I turned it off for 6 hours before trying again. Still no luck. Then I figured it might be a corrupt library somewhere, or something like that, so I formatted and re-installed RH9. I still got the exact same error messages. All I wanted was the aggressive echo cancellation... Now I have nothing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
dtmfmode=inband is evil anyway. :P On Sun, 26 Oct 2003, CW_ASN wrote: I had similar problems, and were related to dtmfmode=inband in sip.conf - Original Message - From: duncan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 9:15 PM Subject: Re: [Asterisk-Users] Asterisk on FreeBSD Is 'top' suggesting that * is actually consuming 98%? If it is, take a look at the * logs for signs of what it might be. We've seen this happen on a lab RH9 system, but its usually while we been doing other unusual things. (In our case, two extra instances of mpg consuming the ~98%; copying *.conf files to a second system that didn't actually have any x100p cards in it, etc.) FWIW, I'm running yesterday's cvs on two RH9 systems just fine. i had a problem with asterisk consuming all the resources available on redhat 9. it would occur roughly every 24 hours or so - and would cause all sorts of problems. when a new channel opened up it fought for resources for a few seconds - so no speech could be heard, then when it could grab enough resources to process the channel it would... but the quality would be terrible. it can be solved with this: export LD_ASSUME_KERNEL=2.4.1 so now thats in my asterisk init script before actually starting asterisk. since doing this i havent had a problem (3 weeks ago). duncna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
Rich Adamson wrote: My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower CPU linux system, Asterisk runs at 0.1% - both without any active channels... Any ideas, anyone recognizing the problem? Is 'top' suggesting that * is actually consuming 98%? Yes, on FreeBSD. If it is, take a look at the * logs for signs of what it might be. We've Can't find anything in the logs, trying to unload modules. FWIW, I'm running yesterday's cvs on two RH9 systems just fine. Asterisk on Linux runs fine here too, but FreeBSD is going mad. Continuing search... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lucent Partner extension to X100P
Hi All... I'm currently trying to have an extension of my Lucent Partner phone system connected to Asterisk using an X100P. The issue I'm having is that the Lucent Partner analog port connection have different ring and dialtone than the one specified for US in indications.conf. Anyone have accomplished this connection before. I need to know what I should use for the different setting in indications.conf. Otherwise, I was able to record the ringing pattern using ztmonitor. How do I analyze this sound file to find out the required information. Thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
On Sun, 2003-10-26 at 08:41, Steve Meyers wrote: On Sat, 2003-10-25 at 18:49, Ken Godee wrote: You did do a make clean first before recompiling? Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk directories and re-checking them out. Then I decided it might be a heat issue, so I turned it off for 6 hours before trying again. Still no luck. Then I figured it might be a corrupt library somewhere, or something like that, so I formatted and re-installed RH9. I still got the exact same error messages. I spoke too soon. After the re-install, I forgot to add fxsks=1 to my /etc/zaptel.conf. Now it works again! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA-186 Registration Issue
Hi y'all, I have an ATA-186 and it seems to be timing out. After allowing sometime to pass, the device seems to disapear, causing requests for extensions directed to the ATA, to default to vmail. Is there something I can do to keep this from happening? Regards, Phillip - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
On Sun, 2003-10-26 at 06:40, CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus How about starting another list: asterisk-ss7 I would also like to see an asterisk-business list where * based solutions providers and telco's and their wannabee versions can find each other to discuss e.g. interconnection, termination and other business opportunities. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Registration Issue
Hi y'all, I have an ATA-186 and it seems to be timing out. After allowing sometime to pass, the device seems to disapear, causing requests for extensions directed to the ATA, to default to vmail. Is there something I can do to keep this from happening? Regards, Phillip Upgrade to the most recent ATA-186 code, which at my last peek was 2.17 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
You see, guys? Some people loves to see masses of messy details and other people don't... What can we do? Regards, Gus - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 6:58 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Man, getting the masses of messy details out in the open is what this is all about. Let it flow and we all get better and can do better! Steve Underwood wrote: OK. If are talking about masses of messy details, then I agree with you. Regards, Steve CW_ASN wrote: Steve: Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations, good... I have no problems with that... We will discuss HERE all the configurations needed to bring up a CCS7 links in ITP, how load a SPC formats, and how can I add an TGCP route in BTS... Sure! Why not? Regards, Gus - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 2:14 AM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch Interesting. Someone thinks that a strategic use for * should be off this list. Someone thought my FAX modem for * should be off this list. However, nobody seems to think a 1000 messages about Grandstream phones should be off this list. Personally I would welcome seeing more of what people are doing in the softswitch area. Regards, Steve CW_ASN wrote: Juan: I think that we must continue with the discussion out of this list. Te contacto por fuera de la lista. Regards, Gus - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:50 PM Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch On Fri, 2003-10-24 at 16:29, CW_ASN - Gus wrote: No, its not 100% accured. * can be used as Softswitch in MGCP... all good softswitchs uses MGCP/TGCP/NCS to manage each endpoint. I have 1 * box under test with Cisco BTS10200, and * works very fine with this softswitch. You could use SIP too... Can you explain that setup a bit more ? You mean that BTS is controling the * box using MGCP or the inverse ? Cause a I have an * box using a BTS+AS5300 as its PTSN gateway using SIP. But the BTS receives the SS7 signaling(via an ITS i think) and controls the AS5300 via MGCP. Then the * box it is another SIP route inside the BTS. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with 2 NIC cards
Hello, I have the quite similiar problem like yours except that both of my NIC have fix public ip from different ISP provider. Unfortunately we are unable to make it work. This is due to some routing issues of the Asterisk box. My collegue was trying hard to seting up the routing tables, but did no succeed. We finally has given up trying. The solution we have make is to have 2 different asterisk iax server and make these server peer to each other, but not yet try though. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:39 AM Subject: [Asterisk-Users] Trouble with 2 NIC cards Greetings everyone. Did anyone try using 2 NIC cards on the machine? For some reason, asterisk can not identify which IP should be used. In the config files (IAX.conf, sip.conf etc), there is a way to bind the IP address but if the machine is hooked to a DHCP server (such as cable modem), then fix IP doesn't work. It should be simple to bind it to a perticular ethernet card (eth0 or eth1) instead of an IP address. Anyone tried multiple NICs with asterisk? Please write your commentes. Thanks. Ricky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Registration Issue
You wouldn't happen to have this, would you? Or know of where I can download it, without signing up w/ Cisco? Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Registration Issue
I just checked on Cisco software center ... There is no 2.17 version ... The latest one, which was released on Oct 22,2003 is 2.16.1, file name : ata18x-v2-16-1-030709a-2.zip Do you need one ? Regards, Alexander http://asterisk.xvoip.com unofficial Asterisk Online Forums Come and share your information. Open your mind to community. - Original Message - From: Phillip Jackson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 26, 2003 8:59 PM Subject: Re: [Asterisk-Users] ATA-186 Registration Issue You wouldn't happen to have this, would you? Or know of where I can download it, without signing up w/ Cisco? Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stealthtele.com = * friendly
Would you be able to send us mock config entries? These would be good for the record. Cheers! Phillip On Oct 26, 2003, at 10:35 PM, Brian West wrote: I just did some testing[sip] with the guys at stealthtele.com with * and everything went great... thinking setting up an account with them sometime soon... He said they were working on IAX but not sure how far out that would be Has anyone else checked them out? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] After start Asterisk, error foung in the messages log file
Hello all After install the Asterisk, setup the configuration file in the sip.conf and extensions.conf and type asterisk with start program. I don't have line card yet. But I found following error in the /var/log/messages. localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: init_module: Input/output error localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: insmod char- major-196 failed (Log) localhost kernel: CSLIP: code copyright 1989 Regents of the University of California localhost kernel: PPP generic driver version 2.4.2 localhost kernel: Zapata Telephony Interface Registered on major 196 localhost kernel: No ISA tormenta card found at d localhost kernel: Zapata Telephony Interface Unloaded localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: init_module: Input/output error localhost insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: insmod char- major-196 failed localhost kernel: CSLIP: code copyright 1989 Regents of the University of California localhost kernel: PPP generic driver version 2.4.2 localhost kernel: Zapata Telephony Interface Registered on major 196 localhost kernel: No ISA tormenta card found at d localhost kernel: Zapata Telephony Interface Unloaded localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: init_module: Input/output error localhost insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: insmod char- major-196 failed (/Log) I cannot start the asterisk. Do yuo have any suggestion for me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stealthtele.com = * friendly
I just did some testing[sip] with the guys at stealthtele.com with * and everything went great... thinking setting up an account with them sometime soon... He said they were working on IAX but not sure how far out that would be Has anyone else checked them out? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail broken voice
Hi, I am trying to set-up voice mail. When I dial the voicemail extension, voice prompt asking for password is braking or intermittent. I see the error File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?!. I am using a grandstream phone. BTW : My calls between extensions and to fwd works fine. Voice prompt at fwd works fine too. Anyone could help me here ? Appreciate Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA-186 Registration Issue
All, -- I've upgraded my ATA-186 successfully to: ata000af453d1a9 Version: v2.16.1 ata18x (Build 030709a) Below is a link to my config, on the ATA: http://www.jacksongrp.com/telephony/dev.htm -- I am utilizing both lines on the Cisco 7960. I have been able to break things on occasion (from recieving calls on ATA - asterisk routes directly to mailbox) by calling one line of the ATA from my Cisco 7960, answering, then blind transfering the call to another line on the ATA. It seems, after doing this, that I can still call out from either of the two lines on the ATA, but I cannot call into those lines - asterisk comedian vmail answers (it's not so funny!) :-) However, I think things may be OK now, as I've upgraded software again, and went through my configs. SO .. can you take a look at my config - how does it compare to yours? AND, have you, also, experienced similar problems? Solutions? Cheers, Phillip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Registration Issue
Utilizing both lines on the Cisco ATA-186! Not, 7960 - it's late! Regards, Phillip On Oct 26, 2003, at 11:45 PM, Phillip Jackson, President CEO wrote: All, -- I've upgraded my ATA-186 successfully to: ata000af453d1a9 Version: v2.16.1 ata18x (Build 030709a) Below is a link to my config, on the ATA: http://www.jacksongrp.com/telephony/dev.htm -- I am utilizing both lines on the Cisco 7960. I have been able to break things on occasion (from recieving calls on ATA - asterisk routes directly to mailbox) by calling one line of the ATA from my Cisco 7960, answering, then blind transfering the call to another line on the ATA. It seems, after doing this, that I can still call out from either of the two lines on the ATA, but I cannot call into those lines - asterisk comedian vmail answers (it's not so funny!) :-) However, I think things may be OK now, as I've upgraded software again, and went through my configs. SO .. can you take a look at my config - how does it compare to yours? AND, have you, also, experienced similar problems? Solutions? Cheers, Phillip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tonight's CVS breaks Grandstream phone
Hi Guys, Tried the disallow=all and allow=all but still getting one way audio with x-lite and messenger. Any update on this problem. Dave - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 24, 2003 1:49 PM Subject: Re: [Asterisk-Users] Tonight's CVS breaks Grandstream phone FYI. Haven't dug enough to be able to report any more, but re-fetched CVS to verify that sometime in the last few days CVS changes now break my GS phone. It appears to be at the RTP level. It seems to set the call up just fine, but no audio is passed back to the instrument. I reverted, and will try to play with this tomorrow unless someone else tells us it's fixed. Thx. B. I am seeing the same error with CVS as of 02:00 today GMT. Grandstream phones will dial, the dialplan will seem to work, but after a few seconds the call fails. Looking at the SIP debug, I see that There was a new feature added last night to allow for codec permission/denial on a per-peer basis in sip.conf. This means that each SIP client can be forced to use particular codecs (at least, that is the intent. more testing, anyone?) So, it seems that the Grandstreams do not elegantly handle some circumstances of codec presentation which were created by these new patches. It is necessary for you to put the following lines in each Grandstream entry in your sip.conf, OR you can put the identical entries in [general] to have it work across all clients. Note that both lines are required: disallow=all allow=all JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-2000 anyone?
If I understand correctly the Sipura people are the same guys that made the Cisco ATA (Komodo phone) or what ever. I'm going to get one of the Sipura SPA-2000 to use and abuse with * I have seen the web interface.. John over at Chagres was nice enough to let me login to one and look around a few weeks back... I'm impressed .. if you guys care to buy one http://www.chagres.net/products/voip/ata.html ... I should have my by next week.. will post more info after that. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help: mixing monitored in out files
Thnx John for your tip, it worked for me. Azher John Todd [EMAIL PROTECTED] wrote: Hi,There is an email in the mail archives from asterisk-users-admin with a script to mix two monitored files. I was trying that script but the wmix always says : Unable to open display "(null)". I am not using X, which wmix requires.Is there any other tool or any help on making a single file out of the in and out monitored files ??TIAAzhersoxmix will probably work for you. It comes as a standard part of the most recent versions of sox.JT___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Exclusive Video Premiere - Britney Spears
[Asterisk-Users] SIPURA SPA-2000 now available
Hey, just thought the list might want to know that SIPURA has released their really cool ATA device, the SPA-2000 If you are interested in purchasing these units we have them cheaper than their own site does :) surf over to http://www.chagres.net/products/voip/ata.html We have also added other stuff, including a Video phone at: http://www.chagres.net/products/voip/phones.html as always volume pricing available john brown VoIP geek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
