Re: [asterisk-users] PJSip CallerID Question
On 4/7/2018 5:50 AM, Daniel Tryba wrote: On Fri, Apr 06, 2018 at 02:27:31PM -0500, Brent Davidson wrote: I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from. I knew how to do that with the old sip format, but can't seem to figure it out with PJSip. So how did you do that? Uhh, I actually don't remember, and I didn't save any of the old config when I upgraded the servers. Yeah, I know. Currently Location A, extension 10 calls Location B, extension 20. CallerID on Extension 20 displays "10" for the callerID. The Desired configuration is for Extension 20 to show "Locati0n B - 10" on the caller ID. I don't want to modify the caller ID for each individual extension as I want the intra-location caller IDs to show just the extension number. (e.g. LocA/Ext. 10 calls LocA/Ext11 - LocA/Ext11's CallerID displays "10", but LocA/Ext10 calling LocB/Ext20 displays "Location A - 10" for caller ID. You examples contradict. Yeah, that was a type. Should read: The Desired configuration is for Extension 20 to show "Location A - 10" on the caller ID. I don't want to modify the caller ID for each individual extension as I want the intra-location caller IDs to show just the extension number. (e.g. LocA/Ext. 10 calls LocA/Ext11 - LocA/Ext11's CallerID displays "10", but LocA/Ext10 calling LocB/Ext20 displays "Location A - 10" for caller ID. Rather than routing these to the "internal" context, should I create another context and somehow parse/manipulate the caller ID in there then route to "internal" ? TIMTOWTDI, but I'd choose to set the CALLERID(name) on the sending side dialplan (where it routes calls to external extensions). Ah. That's probably how I did it before. Not sure why I didn't copy that section of the dialplan over during the upgrade. Probably because I wanted to get incoming calls working at all locations first, then tackle office to office later. Thanks. --- This email has been checked for viruses by AVG. http://www.avg.com WARNING-FRAUDULENT FUNDING INSTRUCTIONS Email hacking and fraud are on the rise to fraudulently misdirect funds. Please call your escrow officer immediately using contract information found from an independent source, such as the sales contract or internet, to verify any funding instructions received. We are not responsible for any wires sent by you to an incorrect bank account. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSip CallerID Question
I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from. I knew how to do that with the old sip format, but can't seem to figure it out with PJSip. For example: Currently Location A, extension 10 calls Location B, extension 20. CallerID on Extension 20 displays "10" for the callerID. The Desired configuration is for Extension 20 to show "Locati0n B - 10" on the caller ID. I don't want to modify the caller ID for each individual extension as I want the intra-location caller IDs to show just the extension number. (e.g. LocA/Ext. 10 calls LocA/Ext11 - LocA/Ext11's CallerID displays "10", but LocA/Ext10 calling LocB/Ext20 displays "Location A - 10" for caller ID. I have my locations set up in the pjsip_wizard.conf file like this: Location A: [Uplink_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30 registration/expiration = 1800 [Location B](Uplink_defaults) endpoint/context = internal remote_hosts = 10.10.20.253:5060 sends_registrations = no accepts_registrations = no sends_auth = no accepts_auth = no Location B: [Uplink_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30 registration/expiration = 1800 [Location A](Uplink_defaults) endpoint/context = internal remote_hosts = 10.10.11.5:5060 sends_registrations = no accepts_registrations = no sends_auth = no accepts_auth = no Rather than routing these to the "internal" context, should I create another context and somehow parse/manipulate the caller ID in there then route to "internal" ? --- This email has been checked for viruses by AVG. http://www.avg.com WARNING-FRAUDULENT FUNDING INSTRUCTIONS Email hacking and fraud are on the rise to fraudulently misdirect funds. Please call your escrow officer immediately using contract information found from an independent source, such as the sales contract or internet, to verify any funding instructions received. We are not responsible for any wires sent by you to an incorrect bank account. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Dropouts During Call
Been doing some more troubleshooting on this issue. Started up TCPDump and let it run for a whole day, then loaded the file into Wireshark. What is interesting is that there doesn't seem to be any lost packets. The RTP sequence numbers are always contiguous. However, if I output the streams as .au files and listen to them there are obvious points where the audio just goes silent in the middle of the person speaking, and it effects both directions. Doesn't make any sense. On 4/4/2018 10:33 AM, Brent Davidson wrote: At the first office, I replaced all the wiring except the in-wall stuff. Checked all the cables to make sure they were correct. I've done cabling for the last 20+ years, so I've usually got a good feel for that. Make all my own cables and do all my own wiring. I still make a habit of checking that first because you never know when somebody is going to decide to swap out a cable with one they just pulled out of hammerspace for one reason or another. All of the duplex and flow control settings are set to auto-negotiate. The switch logs don't show any unexpected amount of collisions, and no receive or transmit errors. I might add that I have the same setup in 8 offices. Right now, only two of the offices are reporting problems. All of these offices were previously operating fine with Asterisk 1.4 installations. Over the past year all offices were upgraded to new phone/fax servers running Asterisk 13. All offices ran fine for several months until the one problem office started having the audio drop-outs, and then a few weeks later the second office started having the same issue. Is there anything in the pjsip code that might cause RTP latency if reverse DNS lookup timed out for one reason or another? On 4/3/2018 5:20 PM, Dave Platt wrote: I looked at your network diagram. Try checking the configuration of the Ethernet ports on the firewall and the Asterisk box. Make sure they are set to auto-negotiate and not set to a fixed speed and fixed duplex. I have found in the past that if one end of a link is expecting auto- negotiation (as the switches probably are) and the other end is expecting a fixed configuration, things can degrade to half-duplex trying to talk to full-duplex, resulting in lots of collisions and packet loss when there is any kind of significant traffic. Your description would be consistent with the firewall introducing lots of LAN collisions when busy, in the central gigabit switch, even if the VoIP traffic isn't passing through the firewall. Also, check the wiring. Check each individual RJ-45 jumper, *and* the in-house wiring, with a proper tester that can verify that the individual pairs are hooked up correctly. I've seen all kinds of hell occur, in situations where somebody used telco-type RJ-45 connecting cables, in place of proper Ethernet connecting cables. The problem is this: in a telco RJ-45 cable (such as was/is often used for proprietary telephone systems) the individual wires are either not in twisted pairs, or are twisted-pairs in a 1-2 3-4 5-6 7-8 arrangement. These work fine for analog connections. They're latent-death-on-wheels for Ethernet. Ethernet only works well if you connect the pairs as a 1-2, 3-6, 4-5, 7-8 arrangement, because this is how the signals are sent electrically. Using the correct connections ensures that the signals on each pair are "balanced" electrically - that is, the two wires in each twisted pair are carrying equal-but-opposite currents for the two sides of an individual signal. This minimizes electrical coupling between pairs, and thus minimizes crosstalk. If you use a telco-style cable (these are often black, and flat), or if you use what looks like an Ethernet cable but which had its wires "punched down" to the connector in the wrong pairing, things go very badly indeed. One twisted pair might be carrying one TX signal and one RX signal. This pretty much *guarantees* terrible cross-talk between the two. The symptoms of this can be as was related... the connection appears to work OK under light load, when there's usually traffic flowing in only one direction at a time. However, when you put a bidirectional load on the connection, the signals going from A to B and from B to A cross-talk with one another, leading to a very high rate of corrupted/dropped packets on the network. This will often show up in the end device's Ethernet packet statistics, if you can get to them... look for a high rate of dropped or "bad" packets, FCS (frame sequence check) errors, etc. I've seen a fair number of cheap "Ethernet" cables that had been manufactured wrong. You should see a color pairing such as http://www.hardwaresecrets.com/how-gigabit-ethernet-works/ indicates - pins 4 and 5 are a pair (blue, and white-and-blue), and the next-outer pins are also a pair (orange, and white-with-orange). If you see a pattern such as "white-with-green, green, white-with-blue, blue
Re: [asterisk-users] Audio Dropouts During Call
At the first office, I replaced all the wiring except the in-wall stuff. Checked all the cables to make sure they were correct. I've done cabling for the last 20+ years, so I've usually got a good feel for that. Make all my own cables and do all my own wiring. I still make a habit of checking that first because you never know when somebody is going to decide to swap out a cable with one they just pulled out of hammerspace for one reason or another. All of the duplex and flow control settings are set to auto-negotiate. The switch logs don't show any unexpected amount of collisions, and no receive or transmit errors. I might add that I have the same setup in 8 offices. Right now, only two of the offices are reporting problems. All of these offices were previously operating fine with Asterisk 1.4 installations. Over the past year all offices were upgraded to new phone/fax servers running Asterisk 13. All offices ran fine for several months until the one problem office started having the audio drop-outs, and then a few weeks later the second office started having the same issue. Is there anything in the pjsip code that might cause RTP latency if reverse DNS lookup timed out for one reason or another? ** On 4/3/2018 5:20 PM, Dave Platt wrote: I looked at your network diagram. Try checking the configuration of the Ethernet ports on the firewall and the Asterisk box. Make sure they are set to auto-negotiate and not set to a fixed speed and fixed duplex. I have found in the past that if one end of a link is expecting auto- negotiation (as the switches probably are) and the other end is expecting a fixed configuration, things can degrade to half-duplex trying to talk to full-duplex, resulting in lots of collisions and packet loss when there is any kind of significant traffic. Your description would be consistent with the firewall introducing lots of LAN collisions when busy, in the central gigabit switch, even if the VoIP traffic isn't passing through the firewall. Also, check the wiring. Check each individual RJ-45 jumper, *and* the in-house wiring, with a proper tester that can verify that the individual pairs are hooked up correctly. I've seen all kinds of hell occur, in situations where somebody used telco-type RJ-45 connecting cables, in place of proper Ethernet connecting cables. The problem is this: in a telco RJ-45 cable (such as was/is often used for proprietary telephone systems) the individual wires are either not in twisted pairs, or are twisted-pairs in a 1-2 3-4 5-6 7-8 arrangement. These work fine for analog connections. They're latent-death-on-wheels for Ethernet. Ethernet only works well if you connect the pairs as a 1-2, 3-6, 4-5, 7-8 arrangement, because this is how the signals are sent electrically. Using the correct connections ensures that the signals on each pair are "balanced" electrically - that is, the two wires in each twisted pair are carrying equal-but-opposite currents for the two sides of an individual signal. This minimizes electrical coupling between pairs, and thus minimizes crosstalk. If you use a telco-style cable (these are often black, and flat), or if you use what looks like an Ethernet cable but which had its wires "punched down" to the connector in the wrong pairing, things go very badly indeed. One twisted pair might be carrying one TX signal and one RX signal. This pretty much *guarantees* terrible cross-talk between the two. The symptoms of this can be as was related... the connection appears to work OK under light load, when there's usually traffic flowing in only one direction at a time. However, when you put a bidirectional load on the connection, the signals going from A to B and from B to A cross-talk with one another, leading to a very high rate of corrupted/dropped packets on the network. This will often show up in the end device's Ethernet packet statistics, if you can get to them... look for a high rate of dropped or "bad" packets, FCS (frame sequence check) errors, etc. I've seen a fair number of cheap "Ethernet" cables that had been manufactured wrong. You should see a color pairing such as http://www.hardwaresecrets.com/how-gigabit-ethernet-works/ indicates - pins 4 and 5 are a pair (blue, and white-and-blue), and the next-outer pins are also a pair (orange, and white-with-orange). If you see a pattern such as "white-with-green, green, white-with-blue, blue, white-with-orange, orange, white-with-brown, brown" where there are four color-matched pairs of wires next to one another, you've got a bad cable. The same error can occur when building wiring is "punched down" to the RJ-45 jacks. A good Ethernet cable-pair tester can spot such things pretty quickly. --- This email has been checked for viruses by AVG. http://www.avg.com WARNING-FRAUDULENT FUNDING INSTRUCTIONS Email hacking and fraud are on the rise to fraudulently misdirect funds. Please call your escrow officer immediately using contract information found
Re: [asterisk-users] Audio Dropouts During Call
Well, I now have another office complaining of the audio drop-outs. Logs are showing the same issues. RTP just stops for awhile then resumes. At the original problem office, I replaced all the network cables, replaced two network hubs, and made sure the phones are all connected correctly. The problem still exists. One thing I did notice during testing is that the audio is perfectly fine as long as there is no internet traffic, but once there is internet traffic, the audio quality drops drastically, then cuts out completely. Once the internet traffic stops, there is about a 2 second lag, then the audio resumes. I find that incredibly odd as we don't use VOIP outside lines, and none of the voice traffic should be passing through our firewall, router, or DSL modem. Internal network traffic, such as moving a file between shared folders on 2 computers on the internal office network does not impact the audio at all. However, if I try to send a file across the VPN, refresh a web page in the browser, or run a bandwidth test from either computer, the audio goes glitchy then drops out until the traffic returns to normal. Attached is a network diagram to show how both offices are set up. There shouldn't be any reason for traffic that goes straight to the internet to affect the internal VOIP traffic. The Asterisk server only runs Asterisk, Hylafax, and a Samba share for the workgroup copier/scanner to save scanned files to. It isn't doing DNS, or anything that would tax it's resources. The servers both have quad-core CPUs and 16 GB of ram. I've tried switching codes between ulaw, alaw, and g.729 and the problem persists at both offices. Any ideas? On 3/26/2018 10:30 AM, Bertrand LUPART - Linkeo.com wrote: Hello, Only one of the servers has the drop-out issues. This location has a network switch on the main desk due to wiring limitations, but several of the other non-problematic offices do as well. I run some offices with similar asterisk configurations, only one experiencing drop-out calls as well. Just visited the impacted office today, discovering their phones are daisy-chained. Still investigating, but i'm pretty confident correctly wiring them on a decent switch will correct the issue. Also double-check the phone are correctly wired (LAN on LAN port and not on computer port) OTOH -- Bertrand LUPART --- This email has been checked for viruses by AVG. http://www.avg.com WARNING-FRAUDULENT FUNDING INSTRUCTIONS Email hacking and fraud are on the rise to fraudulently misdirect funds. Please call your escrow officer immediately using contract information found from an independent source, such as the sales contract or internet, to verify any funding instructions received. We are not responsible for any wires sent by you to an incorrect bank account. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zombie PJSip Channel
I'm having a strange issue with Asterisk 13.17.2 and pjproject-2.7. I have one extension that will occasionally end up in a "Zombie" channel and stop receiving calls. (Note that the console never says "Zombie" it just shows a channel that can't be hung up. Here's an excerpt from a console session showing the problem: cameronpbx*CLI> core show channels concise PJSIP/15-00e9!internal!s!1!Up!AppQueue!(Outgoing Line)!s!!!3!12616!!1520615616.690 cameronpbx*CLI> hangup request all PJSIP/15-00e9 DAHDI/2-1 cameronpbx*CLI> hangup request PJSIP/15-00e9 PJSIP/15-00e9 is not a known channel cameronpbx*CLI> hangup request "PJSIP/15-00e9" PJSIP/15-00e9 is not a known channel cameronpbx*CLI> hangup request DAHDI/2-1 DAHDI/2-1 is not a known channel cameronpbx*CLI> hangup request all cameronpbx*CLI> hangup request all When it gets in this state, I can't even do a "core restart when convenient" as it will sit there and wait forever for that channel to disappear. I have to drop to a command line and do a "systemctl restart asterisk" to get it to reset that extension. The really strange thing is it never happens to any other extensions on the system. Just that one phone. Any ideas? WARNING-FRAUDULENT FUNDING INSTRUCTIONS Email hacking and fraud are on the rise to fraudulently misdirect funds. Please call your escrow officer immediately using contract information found from an independent source, such as the sales contract or internet, to verify any funding instructions received. We are not responsible for any wires sent by you to an incorrect bank account. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift w/ Asterisk 14
Trying to compile app_swift with Asterisk 14.2.1 and getting the
following. Can anybody tell me what I'm missing?:
[root@localhost app_swift-master]# make
____
(_) / __) _
_ ___ _ _ _ _ _| |__ _| |_
( | _ \| _ \ /___) | | | (_ __|_ _)
/ ___ | |_| | |_| | |___ | | | | | | || |_
\_| __/| __/ () |___/ \___/|_| |_| \__)
|_| |_|
gcc -I/opt/swift/include -I/usr/include -g -Wall -fPIC -D_SWIFT_VER_6
-D_AST_VER_14 -c -o app_swift.o app_swift.c
In file included from app_swift.c:33:0:
/usr/include/asterisk.h:300:2: error: #error "Externally compiled
modules must declare AST_MODULE_SELF_SYM."
#error "Externally compiled modules must declare AST_MODULE_SELF_SYM."
^
app_swift.c:34:1: error: expected declaration specifiers or ‘...’ before
string constant
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 304000 $")
^
app_swift.c:34:33: error: expected declaration specifiers or ‘...’
before string constant
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 304000 $")
^
In file included from app_swift.c:36:0:
/opt/swift/include/swift.h:392:1: error: unknown type name ‘swift_voice’
swift_voice * SWIFT_CALLCONV
^
/opt/swift/include/swift.h:405:1: error: unknown type name ‘swift_voice’
swift_voice * SWIFT_CALLCONV swift_port_find_next_voice(swift_port *port);
^
/opt/swift/include/swift.h:415:1: error: unknown type name ‘swift_voice’
swift_voice * SWIFT_CALLCONV swift_port_rewind_voices(swift_port *port);
^
/opt/swift/include/swift.h:426:52: error: unknown type name ‘swift_voice’
swift_voice *voice);
^
/opt/swift/include/swift.h:437:1: error: unknown type name ‘swift_voice’
swift_voice * SWIFT_CALLCONV swift_port_set_voice_by_name(swift_port
*port,
^
/opt/swift/include/swift.h:448:1: error: unknown type name ‘swift_voice’
swift_voice * SWIFT_CALLCONV swift_port_set_voice_from_dir(swift_port
*port,
^
/opt/swift/include/swift.h:459:1: error: unknown type name ‘swift_voice’
swift_voice * SWIFT_CALLCONV swift_port_get_current_voice(swift_port
*port);
^
/opt/swift/include/swift.h:486:55: error: unknown type name ‘swift_voice’
const char * SWIFT_CALLCONV swift_voice_get_attribute(swift_voice *voice,
^
/opt/swift/include/swift.h:502:28: error: unknown type name ‘swift_voice’
swift_voice_get_attributes(swift_voice *voice, swift_params *out_params);
^
/opt/swift/include/swift.h:517:56: error: unknown type name ‘swift_voice’
swift_result_t SWIFT_CALLCONV swift_voice_load_lexicon(swift_voice *voice,
^
app_swift.c:296:1: error: expected identifier or ‘(’ before ‘{’ token
{
^
app_swift.c:461:2: warning: data definition has no type or storage class
[enabled by default]
res = 0;
^
app_swift.c:461:2: warning: type defaults to ‘int’ in declaration of
‘res’ [-Wimplicit-int]
app_swift.c:467:2: warning: data definition has no type or storage class
[enabled by default]
next = ast_tvadd(ast_tvnow(), ast_tv(0, 10));
^
app_swift.c:467:2: warning: type defaults to ‘int’ in declaration of
‘next’ [-Wimplicit-int]
app_swift.c:467:9: error: incompatible types when initializing type
‘int’ using type ‘struct timeval’
next = ast_tvadd(ast_tvnow(), ast_tv(0, 10));
^
app_swift.c:469:2: error: expected identifier or ‘(’ before ‘while’
while (swift_generator_running(ps)) {
^
app_swift.c:596:2: error: expected identifier or ‘(’ before ‘if’
if (alreadyran == 0 && timeout > 0 && max_digits > 0) {
^
app_swift.c:605:2: error: expected identifier or ‘(’ before ‘if’
if (max_digits >= 1 && results != NULL) {
^
app_swift.c:632:2: error: expected identifier or ‘(’ before ‘}’ token
}
^
app_swift.c:634:11: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or
‘__attribute__’ before ‘:’ token
exception:
^
app_swift.c:639:2: error: expected identifier or ‘(’ before ‘if’
if (engine != NULL) {
^
app_swift.c:642:2: error: expected identifier or ‘(’ before ‘if’
if (ps && ps->q) {
^
app_swift.c:646:2: error: expected identifier or ‘(’ before ‘if’
if (ps) {
^
app_swift.c:664:2: warning: parameter names (without types) in function
declaration [enabled by default]
ast_module_user_remove(u);
^
In file included from app_swift.c:45:0:
app_swift.c:664:2: error: conflicting types for ‘__ast_module_user_remove’
ast_module_user_remove(u);
^
/usr/include/asterisk/module.h:342:6: note: previous declaration of
‘__ast_module_user_remove’ was here
void __ast_module_user_remove(struct ast_module *, struct
ast_module_user *);
^
app_swift.c:665:2: error: expected identifier or ‘(’ before ‘return’
return res;
^
app_swift.c:666:1:
[asterisk-users] Audio cut-outs
I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the audio will drop out for between 15 and 20 seconds before suddenly coming back. I've tried running u-Law as the codec and licensed g.729 version 13.0_3.1.7 with exactly the same results. I have tried turning on every logging option I can think of to troubleshoot this but have not been able to find a solution. I'm troubleshooting by remote, so haven't been able to run a wireshark capture yet. pings to the phones from the Asterisk server show no packet loss during the cut-outs. Any ideas? Thanks, *Brent Davidson* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay after Answer
Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: I've faced the same issue. The issue was related to DNS, the reverse lookup query failure caused the delay around(7-9 seconds). The purpose of reverse lookup is to block IP Spoofing attacks. Regards, Faheem On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <br...@texascountrytitle.com <mailto:br...@texascountrytitle.com>> wrote: I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium TDM400 analog card with 2 incoming analog PSTN lines When a call comes in, the system answers, IVR plays, caller dials an extension, Snom 300 rings, handset picked up. Caller continues to hear ringing for another 7 to 10 seconds. Answerer hears a click, a quick burst of audio, then silence, then another click and audio is engaged. I have tried both SIP and RTP debugging and there are absolutely no messages indicating any timeout or retransmit. I am at a total loss. In the past I've always been able to find an answer to issues like this on my own, but this time I just don't know. I was even beginning to suspect the network switch might be bad, but pinging between the server and the phones shows no packet loss and 0.969ms average response time. What am I missing*?* Thanks, Brent Davidson* * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to create channel DAHDI
In trying to troubleshoot the Delay after Answer problem I had before (which seems to be fixed), I have somehow created a new problem: Outgoing calls are now failing with the following message: [Jun 7 13:28:09] WARNING[9247][C-]: app_dial.c:2429 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) But I DO have working dahdi as incoming calls are working correctly. CLI> dahdi show channels Chan Extension Context Language MOH Interpret BlockedIn Service Description pseudo default default Yes 3 mainmenu default Yes 4 mainmenu default Yes CLI> dahdi show status Description Alarms IRQ bpviol CRCFra Codi Options LBO Wildcard AEX410 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) CLI> dahdi show channel 3 Channel: 3 Description: File Descriptor: 14 Span: 1 Extension: Dialing: no Context: mainmenu Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 8 Busy Pattern: 0,0,0,0 TDD: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no HW Gains (RX/TX): Disabled/Disabled SW Gains (RX/TX): 0.00/0.00 Dynamic Range Compression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook CLI> dahdi show channel 4 Channel: 4 Description: File Descriptor: 15 Span: 1 Extension: Dialing: no Context: mainmenu Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 8 Busy Pattern: 0,0,0,0 TDD: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no HW Gains (RX/TX): Disabled/Disabled SW Gains (RX/TX): 0.00/0.00 Dynamic Range Compression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook ** The Hookstates always say offhook for some reason, though I'm not sure why. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium AEX410P analog card with 2 incoming analog PSTN lines Any ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay after Answer
I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium TDM400 analog card with 2 incoming analog PSTN lines When a call comes in, the system answers, IVR plays, caller dials an extension, Snom 300 rings, handset picked up. Caller continues to hear ringing for another 7 to 10 seconds. Answerer hears a click, a quick burst of audio, then silence, then another click and audio is engaged. I have tried both SIP and RTP debugging and there are absolutely no messages indicating any timeout or retransmit. I am at a total loss. In the past I've always been able to find an answer to issues like this on my own, but this time I just don't know. I was even beginning to suspect the network switch might be bad, but pinging between the server and the phones shows no packet loss and 0.969ms average response time. What am I missing*?* Thanks, Brent Davidson* * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10 app_swift problem
I'm trying to get app_swift (app_swift-2.1-b1-ast10) working with Asterisk 10.2.1 on Ubuntu Server 11.10. Everything appears to compile correctly, but when I go to load the module I get the following: server*CLI module load app_swift.so Unable to load module app_swift.so Command 'module load app_swift.so' failed. [Apr 12 13:42:50] WARNING[1200]: loader.c:458 load_dynamic_module: Error loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close [Apr 12 13:42:50] WARNING[1200]: loader.c:848 load_resource: Module 'app_swift.so' could not be loaded. I have installed a licensed version of Cepstral_Allison-8kHz_x86-64-linux_5.1.0 as the voice. Any idea what is going on? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 10 app_swift problem
On 4/12/2012 3:09 PM, Patrick Lists wrote: On 04/12/2012 09:09 PM, Brent Davidson wrote: I'm trying to get app_swift (app_swift-2.1-b1-ast10) working with Asterisk 10.2.1 on Ubuntu Server 11.10. Everything appears to compile correctly, but when I go to load the module I get the following: server*CLI module load app_swift.so Unable to load module app_swift.so Command 'module load app_swift.so' failed. [Apr 12 13:42:50] WARNING[1200]: loader.c:458 load_dynamic_module: Error loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close [Apr 12 13:42:50] WARNING[1200]: loader.c:848 load_resource: Module 'app_swift.so' could not be loaded. Maybe app_swift can not find the Cepstral library? It should show up in the output of the command ldconfig -v. If it can't be found then you should probably add a config file pointing to the location of the Cepstral lib in /etc/ld.so.conf.d (that's on Red Hat. I have no idea about Ubuntu). Regards, Patrick I have /opt/swift in cepstral.conf in ld.conf.so.d. It finds the library just fine on the command line and ldconfig -v gives me: /opt/swift/lib: libswift.so.5 - libswift.so.5.1 libceplang_en.so.5 - libceplang_en.so.5.1 libceplex_us.so.5 - libceplex_us.so.5.1 It almost looks like Asterisk is not seeing the libraries for some reason. I'm trying recompiling Asterisk now that Cepstral is installed to see if that helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO PCI Master Abort (was Re: Help! Logs filling up with errors!)
Well, I was wrong. The messages went away for a day, then came back. I am now rebuilding the server using an older motherboard. Hopefully that will solve the problem. On 12/9/2011 4:09 PM, Brent Davidson wrote: For the sake of posterity, I'm posting this solution: When I checked the server, the PnP OS option in the BIOS was set to No. Changing the option to Yes and rebooting has solved the problem. On 12/8/2011 10:58 AM, Brent Davidson wrote: I am still having issues with the error message Dec 7 14:25:06 servername kernel: FXO PCI Master abort filling up my log files. I've temporarily managed a work around by having the message log emptied every 10 minutes, but this is not a permanent solution. I expanded my google search to simple kernel pci master abort and came across a couple of sites recommending that the BIOS option PnP OS be set to No to solve these problems. Does anyone have any experience with this and think this might actually help? (The problem server is in a remote office and I don't want to make the 2 hour drive until I'm sure I have a solution.) Thanks, Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO PCI Master Abort (was Re: Help! Logs filling up with errors!)
For the sake of posterity, I'm posting this solution: When I checked the server, the PnP OS option in the BIOS was set to No. Changing the option to Yes and rebooting has solved the problem. On 12/8/2011 10:58 AM, Brent Davidson wrote: I am still having issues with the error message Dec 7 14:25:06 servername kernel: FXO PCI Master abort filling up my log files. I've temporarily managed a work around by having the message log emptied every 10 minutes, but this is not a permanent solution. I expanded my google search to simple kernel pci master abort and came across a couple of sites recommending that the BIOS option PnP OS be set to No to solve these problems. Does anyone have any experience with this and think this might actually help? (The problem server is in a remote office and I don't want to make the 2 hour drive until I'm sure I have a solution.) Thanks, Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO PCI Master Abort (was Re: Help! Logs filling up with errors!)
I am still having issues with the error message Dec 7 14:25:06 servername kernel: FXO PCI Master abort filling up my log files. I've temporarily managed a work around by having the message log emptied every 10 minutes, but this is not a permanent solution. I expanded my google search to simple kernel pci master abort and came across a couple of sites recommending that the BIOS option PnP OS be set to No to solve these problems. Does anyone have any experience with this and think this might actually help? (The problem server is in a remote office and I don't want to make the 2 hour drive until I'm sure I have a solution.) Thanks, Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help! Logs filling up with errors!
I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel sources. Hardware is 2 X100P Wildcards. Everything seems to be working OK but my logs are filling up with this message: Dec 7 14:25:06 servername kernel: FXO PCI Master abort The messages just pour in constantly until the hard drive is full. It's eaten 50+ gigs 4 times already today. OS is Centos 6.0, with custom kernel is 2.6.39.4.smp.x86_64. The motherboard is an Asus M4A78LT-M LE running a dual-core AMD CPU and 4gigs of ram. Does anyone know what might be causing this? Thanks, Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! Logs filling up with errors!
On 12/7/2011 2:35 PM, Danny Nicholas wrote: Check this post - it sounds like exactly what is happening to you. http://osdir.com/ml/debian.packages.voip.devel/2009-01/msg00046.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, December 07, 2011 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help! Logs filling up with errors! I am running Asterisk 1.6.2.20 with dahdi 2.5.0.2, Oslec from kernel sources. Hardware is 2 X100P Wildcards. Everything seems to be working OK but my logs are filling up with this message: Dec 7 14:25:06 servername kernel: FXO PCI Master abort The messages just pour in constantly until the hard drive is full. It's eaten 50+ gigs 4 times already today. OS is Centos 6.0, with custom kernel is 2.6.39.4.smp.x86_64. The motherboard is an Asus M4A78LT-M LE running a dual-core AMD CPU and 4gigs of ram. Does anyone know what might be causing this? Thanks, Brent -- Yes, that appears to be what is happening to me, but I can't seem to find a solution. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Provider Recommendation in US
I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fail2ban - SuSEfirewall
The problem sounds like fail2ban is failing to write the new rules to a permanent file, which would otherwise allow the rules to persist after a reboot. Tilghman, That is exactly right. I'm thinking I need to revise the SuSEfirewall init scripts to follow up with restarting fail2ban, but then I think fail2ban will need to have a persistent jail after restarting, which I did find online. I am a big fan of centralized management, so I prefer to do that rather than have static IP addresses on the network (except of course where absolutely essential). For the OP: maybe a workaround is to assign a fixed IP address from your DHCP server and use a very long lease time? John, Agreed re management. The lease would have to be real long, like a year or so. That would do the trick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fail2ban - SuSEfirewall
I have tried to setup fail2ban on a machine running OpenSuSE 11. Everything looks fine, except the machine restarts the firewall whenever the DHCP lease is renewed, thus flushing all the fail2ban rules (I think.). It seems to me that a quick fix would be to have the system restart fail2ban whenever the firewall is restarted. Has anyone else encountered this issue? .and come up with a solution? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring splash
Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. On 5/26/2010 11:36 AM, Jeff LaCoursiere wrote: Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- Brent Davidson Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 br...@texascountrytitle.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring splash
On 5/26/2010 1:16 PM, Tim Nelson wrote: - Jeff LaCoursierej...@jeff.net wrote: On Wed, 26 May 2010, Brent Davidson wrote: Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. Now that sounds like a good plan. But a quick look through the options in zapata.conf don't show any kind of option for waiting before pickup. It would be in your dialplan. (Untested, OTMH, etc) Dialplan: [from-analog-lines] exten = s,1,Wait(2) exten = s,n,Answer() exten = s,n,Play(tt-monkeys) exten = h,1,Hangup() Again, that is untested, just off the top of my head. The key is putting a wait before your Answer(). A phantom ring/ring splash should fade away before the Wait() period is finished, therefore not hitting your Answer() or Dial() or whatever you have causing all sorts of panic and grief. :-) --Tim I was thinking there was a way to directly set the number of rings before the system picked up the call, but it looks like Tim is right. The Wait statement before the answer appears to be the only way to handle this. I actually used this technique to deal with some phantom rings that were occurring at one of my branch offices. The Telco had the switch set up to periodically test the line (like every 30 minutes) and Asterisk was detecting those test pulses as a ring and answering the call, then passing it on into the operator queue before the system could detect the hang-up. The poor lady at that office nearly had a nervous breakdown before I figured out how to filter out the phantom calls with the wait command. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SuSE Firewall2 - Port Forward Command
Does anyone know what commands in the config file for a SuSE Firewall will forward 5060 and RTP ranges to an Asterisk box in the internal LAN? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 4/7/2010 2:45 AM, asterisk card support wrote: hi: how about the codecs? Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com I have the phones and asterisk limited to ulaw only. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 3/31/2010 10:38 AM, Michael L. Young wrote: Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) In my case, no. All extensions are hard-coded. We only have a handful of phones that don't change. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 3/31/2010 12:06 PM, Danny Nicholas wrote: Just to get a 100% correct response to last question, are you using the flat CDR or mysql/some other DB? All sip clients/peers are defined in sip.conf, dial-plan is entirely in extensions.ael. We have one office that uses an Asterisk native database call in the dialplan for the operator extension to see which extension is currently handling operator calls, but other than that there is no no DB used on any of the other systems. -Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropped Calls
I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 3/30/2010 3:14 PM, Danny Nicholas wrote: A few thoughts; 1. I assume that the * servers aren't on dedicated networks; Do the dropped or one-way calls occur during high-traffic times or are they concurrent with large downloads? In my shop, we had to get a router that would prioritize voice traffic or we would be dead in the water during client file transmissions. Asterisk servers are not on a dedicated network, but our total network utilization is less than 10% max at any time. 2. Don't know about the SNOM or GS phones, but my Polycom phones let you establish higher packet priorities for voice traffic as well. I have all the phones, the asterisk server and the core switch set to prioritize RTP and SIP packets at top priority. But I never see any indication of dropped or delayed packets in the logs. 3. Have you been able to do a top during one of these failures? Could be a memory leak that comes up randomly. This one is a tough one. When these types of calls occur it is completely random. Sometimes there will be one or two in a row, other times there won't be one for a couple of days. It would take some some serious logging to catch top data at the exact moment one of the calls drops or the one-way audio hits. 4. Looking at the startup logs, are the cards having to retry several times to get an IRQ? Digium cards IME can conflict with the Hard Drive (SCSI) controller, causing problems during heavy I/O periods. Hope this helps Cards all get an IRQ on the first try. Other data of interest: Our main office only has 8 incoming analog lines, the other offices all only have 2 incoming lines, and there is no correlation between calls in progress and and either of the problems. Sometimes the main office will have two or three in-progress calls and another incoming or outgoing call will experience one-way audio or a disconnect and the others are unaffected. Not even a glitch in the audio. I have had both problems happen to me after hours when I was the only one in the office so the network was completely idle and my call was the only one active. I've been trying to trace this problem for about two years and still have not been able to make any real progress. I guess I should just update to Dahdi and Asterisk 1.6, but I just hate to change a system that is (mostly) working. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One-Way Audio after Hold
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though the local phone can hear the Sipphone user. Every possible combination of nat=yes, no, maybe, possibly or never gives the same result. Further, canreinvite=yes/no/nonat has no result. I suspect a possible reinvite issue with Asterisk being out of the RTP stream, so I have tried all the usual variables in the DialI() command as well to no avail. Any thoughts on how to fix one-way-audio after a hold? --Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP Kall One-Way Audio
I've scoured the web for hints, and find a lot of chatter about one-way audio with IP Kall, but no definitive explanation. I have the default range (5000-31000) of UDP RTP ports forwarded right to my asterisk box, and have no other difficulties with one-way audio on any other peers. Does anyone know of a special setting or issue with IP Kall? --Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
SPA504G - 1 more vote for it. It is worth having 4 lines even if you need 1 initially. SPA504G supports G722 and sound is awesome even if you do not not use teh HD sound. If you do not care that mcuh about HD sound and do not need PoE SPA941 is a excellent choice - you get really a lot for the price Peter Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced SIP compatibility with asterisk more completely with the SPA504G's than they have the 7940 series? Lack of features on the 7940's is frustrating, and makes me hesitant to try other Cisco phones, even if the SPA504G is newer. --Brent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sporadic one-way audio
We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear us. If we immediately call them back everything is fine. The problem affects all offices and also happens when making sip to sip calls from one snom 300 to another. In addition we periodically have calls that drop off in the middle of a conversation like the connection was lost. I haven't been able to replicate any of these problems and the people that are having them can't seem to keep track of when they occur so I can go back and look in the logs. I suspect that both problems may be related though. Possibly a registration issue? Any ideas are welcome. Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting Transfer
Is there a way to detect if a call is a transfer in the dialplan? Here
is my issue: I have an office with 2 extensions. Under normal
circumstances any call that comes in should ring both extensions. I
accomplish this through a queue. The problem is that if the call is
answered on say extension 11 and the answerer wants to transfer the call
to the other phone, extension 10, transferring the call to extension 10
puts it back in the queue that again rings both phones. I want to set
the system up so that if the call is a transfer from the other extension
it will only ring the phone it's being transferred to. This is what I'm
currently doing (using AEL dialplan):
10 = {
if (${CALLERID(num)} = 11) {
internal-ext(${EXTEN},SIP/${EXTEN});
} else {
Queue (operator|tTnHr|||30);
}
Voicemail(1...@internal|u);
Hangup;
}
11 = {
if (${CALLERID(num)} = 10) {
internal-ext(${EXTEN},SIP/${EXTEN});
} else {
Queue (operator|tTnHr|||30);
}
Voicemail(1...@internal|u);
Hangup;
}
My only problem is that we have some extension duplication at other
offices and it is possible for an extension to come in from another
office with the same CallerID Number.
Is there a better way to do this?
Thanks,
Brent
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[asterisk-users] Russia Calls Skype/VoIP Security Threat
Anybody seen this article yet? Looks like Russian Telecom business have decided that VoIP is going to put a dent in their profits so their pitching it as a threat to Russia's national security and working to get laws put into place to make sure the government controls VoIP providers operating in or providing services to Russia. http://www.reuters.com/article/technologyNews/idUSTRE56N41I20090724?feedType=RSSfeedName=technologyNewsrpc=22sp=true ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hunt group
Gondar Monn wrote: I am having trouble with a DID on a PRI. If there is a call to that DID (let say 5551234) , the next calls get a busy signal. How to I go about sending the call to the next available channel ? Thanks! G. If the telco is providing the PRI then you need to tell them you want rollover on the PRI's. Otherwise, anybody calling across the PSTN to the DID number that is bound to the PRI channel is going to get a busy signal from the telco if that channel is in use. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfer option and pressing #
Alex Samad wrote:
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
My problem is that when you call something like internet banking they
want #, but when # is pressed asterisk gets it instead. is there a way
around this ?
I haven't been able to get asterisk to listen to flash either
Alex
The easiest solution would probably be to look in features.conf and
change the option for forwarding to require two consecutive # presses.
The other option would be to put an explicit dial rule for the numbers
that need the # bypass and have them omit T and from the dial command.
You could also set up a dat abase with a simple web front end for your
users to enter numbers that need to have the transfer function bypassed
and do something like this (I use AEL so this is in AEL Format)
macro specialDial (ext) {
if (${DB_EXISTS(bypass/${ext})}) {
Dial (${TRUNK}/${ext});// Dial without transfer
} else {
Dial (${TRUNK}/${ext},,T); // Dial With Transfer
}
}
This is assuming you create a table called Bypass in your Asterisk
Database and add the number to the database.
Good luck,
Brent
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Re: [asterisk-users] Automatic Gain Control
Danny Nicholas wrote: If you are using a large number of DAHDI channels, you could designate a chunk of them as non-local since you can control RXGAIN on each channel. You would have to work out something with your TELCO since your'e a dead duck control-wise once you answer the call. Yuck. I could see that being a temporary workaround, but it is not a good permanent solution. And even as a workaround it wouldn't work for my application. Each of our remote offices normally only has 1 employee (2 at most) and 2 incoming lines in a rollover setup. I know I've probably asked this before but which parameters do txgain and rx gain control? I've heard conflicting explanations. Looking at it from a telco equipment standpoint I would say rxgain should be the gain on the sound received from the far end of the PSTN and txgain is the sound leaving the TDM card over the PSTN. But I've seen a couple of explanations say that rxgain sets the volume of sound flowing into the zap/dahdi module from other channels and that txgain sets the volume flowing out of the zap module to other modules. That would have the effect of reversing what seems like logical functions and make rxgain actually control the volume being sent out to the PSTN and txgain set the volume coming in from the PSTN. I have not had opportunity to run any tests to verify for myself which explanation is correct. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic Gain Control
Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8 solves the problem with long-distance lines, but then local calls say the person in my office is too loud. I understand that it is going to be difficult to reliably detect a major drop in the volume at the far end of the call, but I'm just wondering if there is a good solution for this. We're using Rhino WC4-FXO-ec cards and the OSlec echo canceler (since the on-board echo canceler didn't seem to help our echo issues) Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
Julien Claassen wrote: Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the misdn send digit command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a completely different mechanism? Any help here is appreciated. A pointer to the right part of the documentation is completely sufficient. Warm regards Julien Putting a person on hold using DTMF is part of the feature code mechanism. You configure it in features.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] false answer on zaptel
Botond Botyanszki wrote: Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. I've been digging google, wikis but have not found what might be causing this. SIP works fine, this problem seems to be only zaptel specific. I could use the NVLineDetect application but I think this would be a hack around the problem. Before I start fixing the nvlinedetect code so that it compiles and works with asterisk 1.4 I thought I should ask here first. Any suggestions? Thanks, Botond What Telco are you using? Do you have callprogress=yes or hanguponpolarityswitch=yes in your zapata/dahdi .conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speex problem installing on CentOS 5.3
Steve Totaro wrote: On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote: John A. Sullivan III wrote: Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log configure:43813: checking for speex_encode in -lspeex configure:43848: gcc -o conftest -g -O2 conftest.c -lspeex -lm 5 configure:43906: checking speex/speex.h usability configure:43947: checking speex/speex.h presence configure:44015: checking for speex/speex.h configure:44076: checking for speex_preprocess_ctl in -lspeex configure:44111: gcc -o conftest -g -O2 conftest.c -lspeex -lm 5 /home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined reference to `speex_preprocess_ctl' | #define HAVE_SPEEX 1 | #define HAVE_SPEEX_VERSION | char speex_preprocess_ctl (); | return speex_preprocess_ctl (); configure:44341: checking for speex_preprocess_ctl in -lspeexdsp configure:44376: gcc -o conftest -g -O2 conftest.c -lspeexdsp -lm 5 /usr/bin/ld: cannot find -lspeexdsp | #define HAVE_SPEEX 1 | #define HAVE_SPEEX_VERSION | char speex_preprocess_ctl (); | return speex_preprocess_ctl (); Internet searches have only further confused the issue for me. It seems this is part of libspeex which in the RedHat world is provided by the speex-devel package (which I have installed): [r...@pbx01 ~]# rpm -qa | grep speex speex-devel-1.0.5-4.el5_1.1 speex-1.0.5-4.el5_1.1 What is the magic to make speex available to Asterisk on CentOS 5.3? Or am I stuck having to uninstall the speex packages and install speex from source? Thanks - John I ended up having to install from source. There are apparently bits of speex that are not included in the RPM's. It's a farily simple install though. Good luck, -Brent I am curious if a yum -y install speex* would have worked for you? I will give it a try on my next 5.3 box. That was the first thing I tried before trying yum -y install speex-devel There was always some link or library missing or possibly just in a non-standard location. Installing from source I just did a configure, make, and make install then all was good. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speex problem installing on CentOS 5.3
John A. Sullivan III wrote: Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log configure:43813: checking for speex_encode in -lspeex configure:43848: gcc -o conftest -g -O2 conftest.c -lspeex -lm 5 configure:43906: checking speex/speex.h usability configure:43947: checking speex/speex.h presence configure:44015: checking for speex/speex.h configure:44076: checking for speex_preprocess_ctl in -lspeex configure:44111: gcc -o conftest -g -O2 conftest.c -lspeex -lm 5 /home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined reference to `speex_preprocess_ctl' | #define HAVE_SPEEX 1 | #define HAVE_SPEEX_VERSION | char speex_preprocess_ctl (); | return speex_preprocess_ctl (); configure:44341: checking for speex_preprocess_ctl in -lspeexdsp configure:44376: gcc -o conftest -g -O2 conftest.c -lspeexdsp -lm 5 /usr/bin/ld: cannot find -lspeexdsp | #define HAVE_SPEEX 1 | #define HAVE_SPEEX_VERSION | char speex_preprocess_ctl (); | return speex_preprocess_ctl (); Internet searches have only further confused the issue for me. It seems this is part of libspeex which in the RedHat world is provided by the speex-devel package (which I have installed): [r...@pbx01 ~]# rpm -qa | grep speex speex-devel-1.0.5-4.el5_1.1 speex-1.0.5-4.el5_1.1 What is the magic to make speex available to Asterisk on CentOS 5.3? Or am I stuck having to uninstall the speex packages and install speex from source? Thanks - John I ended up having to install from source. There are apparently bits of speex that are not included in the RPM's. It's a farily simple install though. Good luck, -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN Connection
Lyle Giese wrote: Manoj Panicker - FOES wrote: Hi Which is the best interface card to connect* PSTN* line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj That's a wide open question. How many lines? What kind of lines? What country are you in? What options are availible to you? I only have three incoming lines for a soho Asterisk install. I decided on a T1 card and picked up a used channel bank on ebay. Not the cheapest way, but it has served me very well. You are not going to get much help unless you define the problem better. Lyle Giese LCR Computer Services, Inc. HI, OK, I'm going to chime in on this one as I am going to set up an Asterisk system for our volunteer ambulance service. As a part of the Emergency Services we need to maintain a POTS line as redundancy and due to the fact that with an old style phone I don't need power for the phone to work. I plan on using a SIP provider for the rest of our phone needs. If not for the emergency services part I would go completely SIP based. Anyway I would need a FXO/FXS card for use in the US. Only one line so I don't need any of the fancy 4 line systems. I have heard you can use certain modems to do this but I would like what I am doing to be seamless and not require hacking at a problem for hours to save $50. I just want it to work quick and easy. I am unsure what you mean by What kind of lines? and What options are availible to you?. Maybe that is part of asking this question, to get some info about the phone system too. Any help would be grand. Thanks Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Page/Intercom problem
openSuse 11 Asterisk 1.4.23.1 Asterisk GUI 2.0 Latest SVN version I set up some page groups using the Asterisk GUI and found that when I hang up the paging phone it causes Asterisk to restart. So far no one has been on the phone at this time so I am unsure if it hangs them up but it definatly drops me out of the CLI back to the Linux command line and it restarts the GUI interface when it happens. This is not an intermediate type thing it happens every time. Anyone have a clue as to what is going on or how I should troubleshoot this? Thanks Brent -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Recognition
Have you tried relaxdtmf=yes in zapata.conf/dahdi.conf? -Brent Timm M.Schneider wrote: Hi, is there a possibility to tell zaptel or Asterisk to modify the DTMF sensibility? The problem what i have is that the Asterisk don't get all Numbers which the analog-FAX dial, let say the FAX dial 123456789 the Asterisk get to number 24679. I think that can be to DTMF Tone duration or the Frequenzy. so you got yna idea what it could be? Thx for helping me. Bye Timm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked Calls Problem
openSuse 11 Asterisk 1.4.23.1 Asterisk GUI 2.0 When parking a call it does not tell me what extension it parked the call on. I think I read something in the mail list that mentioned a problem with call parking and one of the Asterisk 1.4s. Is 1.4.23.1 one of those version having issues? Thanks -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF received twice
Administrator TOOTAI wrote:
David fire a écrit :
out there is a file to change the dtmf duration
where are you?
France
[...]
from other phones like lkand lines it works well?
No, the same. The called number is a number received by a trunk SIP, the
GW is also setted as dtmfmode=auto. Calling from mobile phone or
landline to other services using DTMF -like banks- is OK.
I make further tests and so that setting dtmfmode=info for this GW make
DTMF working correctly! Is this the normal behaviour?
Our dialplan works great for others GW's, if this is normal we have to
adapt it in case of dtmfmode=info. From where can we get the dtmf type?
For me it looks like a bug.
Thanks for your help.
2009/5/11 Administrator TOOTAI ad...@tootai.net
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the one of the calling mobile phone
exten = s,n,Background(silence/1)
; Nokia E65 send digits in DTMF mode, no need to take care about input
corrections
;
exten = s,n(enterDigits),Read(myExten,pls-entr-num-uwish2-call,0,,,3)
exten = s,n,GotoIf($[${myExten}=]?enterDigits)
[...]
Problem is that received DTMF digits in ${myExten} are received twice eg
for 1234 ${myExten} has 11223344. I correct the extension by dialplan
but I think it's not really a solution.
In sip.conf, the dtmfmode is set to auto. If I set it to rfc2833, the
same behaviour.
Can somebody confirm this before I open a bug, thanks.
Regards
--
Daniel
I've seen a couple of examples of this on the list where a provider
sends DTMF in multiple formats and Asterisk with dtmfmode=auto picks up
all the digits sent in all formats. Maybe there should be a code change
so that dtmfmode=auto makes asterisk lock on to the mode of the first
digit received for a session and ignores all other formats for that
particular session? Does that make sense to anybody?
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Re: [asterisk-users] QoS VPN
David Backeberg wrote: On Thu, May 7, 2009 at 3:54 PM, Brent Davidson br...@texascountrytitle.com wrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. VNC is very asymmetric. It doesn't generate much traffic from the person viewing, and it generates lots of traffic FROM the system being viewed. This helps explain why the system being viewed side can hear incoming voice packets, and outbound voice packets that have to compete with the large amount of outgoing video signal data lose. QoS may or may not help you here. Well, the fact that our central office has a 10mb downstream / 5mb upstream connection (Two 5Mb down 2.5Mb up DSl connections load shared) helps with them hearing me clearly too, I'm sure. I can get the packets to them faster than they can get packets to me. If voice quality is important, you should have a separate connection dedicated to just voice. The obvious workaround is grab your cell phone and call them with that. You DO have a way to dial directly to that office without going over the PIX, right, right? How do you call the remote office when the PIX goes down? What will help you is getting a bigger line or separating the voice traffic from the data traffic completely. If you are good with ssh, you can also do a compressed ssh tunnel to encrypt and on-the-fly compress the VNC session. But if this is Windows good luck with that. Yes, we can dial all satellite office through the PSTN if we really want to, but one of the reasons we went to a VOIP system was to cut down on the long-distance charges that result from office-to-office calls, and to be able to transfer calls from one office to another. All in all the system works as designed, except for the rare occasions that I'm doing support with VNC and have a person on the remote extension as well. But just because nobody else has complained yet doesn't mean there aren't other conditions that could trigger a poor-quality call. If I can find a solution that works in my worst-case VNC situation then maybe I'll prevent a few future issues from ever becoming real problems. Separating the voice off to it's own connection would defeat the cost-cutting reasoning behind the system. Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
Jeremy Mann wrote: Access-list 100 permit ip host asterisk server any Class-map match-any voip Match access-group 100 Policy-map voip Class voip Priority 256 Class class-default Fair-queue Interface fastethernet 0 Service-policy output voip Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls, adjust accordingly. You must also use the qos pre-classify in your ipsec tunnel definitions for this to work, but it does work well. I know I'm potentially mapping other traffic than voip, but I'm lazy and don't want to classify the rtp and sip and iax ports, rarely does the box do any other traffic than voip as updates occur in off hours. You'll probably additionally want to match your ipsec keying traffic and give it priority bandwidth, if you're going to push voip through the tunnel you'll find yourself rekeying more often and want to make sure on a saturated link it gets priority so the tunnels don't drop. If you're on DSL, you probably want to research cascading the Qos, have a root policy that throttles all bandwidth to a certain speed, then a child policy that prioritizes that bandwidth, so you don't saturate your outbound circuit(think in terms of P2P protections). Thank you. This is EXACTLY what I was looking for. Do the packet counters for show policy-map int fast 0/0 only increment when the queuing kicks in or should they be incrementing all the time as packets flow? Thanks again, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Master.csv
Hello, I am getting the following error on my CLI [May 6 15:59:20] ERROR[25789]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory I am a bit of a Linux newb so please be gentle. I assume this has something to do with the fact that there are two slashes between asterisk//cdr-csv and cdr-csv//Master.csv I have looked at all the .conf files that deal with CDR and cannot find the entry for this file location. logger.conf and asterisk.conf have not born fruit either. /var/log/asterisk/cdr-csv does not exist and this Master.csv does not exist. Running Asterisk 1.4.23 on openSuse 11. I am also using the Asterisk GUI 2.0 for my interface. Thanks in advance for any help. -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Master.csv
Thanks to everyone for the help. I suppose questions this easy to answer can be a nice diversion, at least they are for me. I thought it might be as easy as adding the directory but the double slashes // in the CLI error message threw me off. Anyway adding the directory worked and I am now getting the CDR logged. I don't know if I really need them but I have them. So last question on this. Why are there double slashes in the CLI error message? Thanks again Brent Danny Nicholas wrote: Here is your problem. The directory /var/log/asterisk/cdr-csv must exist for asterisk to write it's plain-jane (their term) text CDR file. This is defined in cdr.conf (it's the last working section of mine). You can create the directory or comment out that section of cdr.conf. Your choice. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Vrieze Sent: Thursday, May 07, 2009 8:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Master.csv Hello, I am getting the following error on my CLI [May 6 15:59:20] ERROR[25789]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory I am a bit of a Linux newb so please be gentle. I assume this has something to do with the fact that there are two slashes between asterisk//cdr-csv and cdr-csv//Master.csv I have looked at all the .conf files that deal with CDR and cannot find the entry for this file location. logger.conf and asterisk.conf have not born fruit either. /var/log/asterisk/cdr-csv does not exist and this Master.csv does not exist. Running Asterisk 1.4.23 on openSuse 11. I am also using the Asterisk GUI 2.0 for my interface. Thanks in advance for any help. -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QoS VPN
I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PA system with cepstral
Alternatively look into the M() option to Dial to execute a Macro upon connect. You could have your macro setup to call the cepstral app. -Brent Justin Killen wrote: That works great -- Thanks Danny! -Justin *From:* Danny Nicholas [mailto:da...@debsinc.com] *Sent:* Monday, April 20, 2009 12:23 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Asterisk PA system with cepstral Here is one way Create a call file Channel: SIP/100 CallerID: SIP/104 MaxRetries: 1 WaitTime: 60 retryTime: 5 Application: background Data: /tmp/systemisup Have your dialplan create and send the call file for each person you want to get the Cepstral file. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Justin Killen *Sent:* Monday, April 20, 2009 2:14 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk PA system with cepstral Hi All, We just bought a sip based PA setup here with the intention of hooking it into our existing asterisk (1.4) setup. It works as expected when I dial it's extension, but I want to have system generated speech played based on some action (using cepstral, which is already installed and working). My first thought was to DIAL to the extension, and then have cepstral play the audio. The problem with this is (of course) that once the dial connects, the sound doesn't get played until after a hangup. My next thought is to create a conference call between the console and the PA, but I'm not sure how to initiate the call on the pbx side and then play audio onto the line. Thanks in advance -Justin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
To the best of my knowledge, the only way for you to control the duration sent to the PSTN lines is for you to be directly connected to the lines so you can set the tone duration in zapata.conf / dahdi.conf or to use inband signalling. One thing you might try is researching the SipDtmfMode command. It allows you to change the DTMF mode on an active channel. A suggestion might be to set up the dial command with the M() option that point to a Macro that changes the DTMF to INBAND once you are connected to the problem number. At least in theory, if your provider is expecting RFC2833 and they get inband, they should just ignore the inband signaling and pass it on as part of the audio stream. The only problem is that this may only work if you use uLaw or aLaw for your codec and I don't know exactly how to set the tone duration without having a zapata.conf or dahdi.conf entry. Even with one of those files, I don't know how Asterisk chooses to do the rfc2833 to inband translation or where it pulls the toneduration setting from if no PSTN interface is involved in the call. -Brent John covici wrote: OK, thanks. If I could convince them to use info, would that be better as far as the duration is concerned? on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
It's been around awhile. I've used it in 1.4 Check out this link for basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode John covici wrote: Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. Is this new in 1.6? on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote To the best of my knowledge, the only way for you to control the duration sent to the PSTN lines is for you to be directly connected to the lines so you can set the tone duration in zapata.conf / dahdi.conf or to use inband signalling. One thing you might try is researching the SipDtmfMode command. It allows you to change the DTMF mode on an active channel. A suggestion might be to set up the dial command with the M() option that point to a Macro that changes the DTMF to INBAND once you are connected to the problem number. At least in theory, if your provider is expecting RFC2833 and they get inband, they should just ignore the inband signaling and pass it on as part of the audio stream. The only problem is that this may only work if you use uLaw or aLaw for your codec and I don't know exactly how to set the tone duration without having a zapata.conf or dahdi.conf entry. Even with one of those files, I don't know how Asterisk chooses to do the rfc2833 to inband translation or where it pulls the toneduration setting from if no PSTN interface is involved in the call. -Brent John covici wrote: OK, thanks. If I could convince them to use info, would that be better as far as the duration is concerned? on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-beginner : cannot make phonecallsusingAsterisk
Danny Nicholas wrote: Do you have include=intern in the default context? If no, * will come back with can't find peer 210 (or 211). *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jonas kellens *Sent:* Monday, April 13, 2009 11:19 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : /[r...@asterisk asterisk]# cat sip.conf/ /[general]/ /bindport=5060/ /bindaddr = 0.0.0.0/ /[BT201]/ /type=friend/ /context=intern/ /host=192.168.4.210/ /secret=testpaswoord/ /[GXP1200]/ /type=friend/ /context=intern/ /host=192.168.4.211/ /secret=testpaswoord/ extensions.conf : /[r...@asterisk asterisk]# cat extensions.conf/ /[intern]/ /exten = 210,1,Dial(SIP/BT201)/ /exten = 211,1,Dial(SIP/GXP1200)/ Asterisk CLI shows me : /asterisk*CLI sip reload/ /Reloading SIP/ / == Parsing '/etc/asterisk/sip.conf': Found/ / == Parsing '/etc/asterisk/users.conf': Found/ / == Parsing '/etc/asterisk/sip_notify.conf': Found/ /asterisk*CLI sip show peers/ /Name/username HostDyn Nat ACL Port Status / /GXP1200192.168.4.211 5060 Unmonitored / /BT201 192.168.4.210 5060 Unmonitored / /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]/ /asterisk*CLI dialplan show intern/ /[ Context 'intern' created by 'pbx_config' ]/ / '210' = 1. Dial(SIP/BT201) [pbx_config]/ / '211' = 1. Dial(SIP/GXP1200) [pbx_config]/ I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This is not the case since both of his phones are configured to come in to the intern context by default. In the real world, if you intern context had access to outside calls and you included it in the default context and happened to allow guest access, then anybody coming in to your box could make outbound calls. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOS Interface
Jorge Mendoza wrote: Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ You're comparing to apples to Orange. IOS is the Cisco operating system run by the Femtocells, not the protocol. I'm not familiar with Femtocells, but as far as I can tell (from reading wikipedia) they apparently can do SIP internally, but that is a more advance configuration and might require some additional software. Looks like they are more designed to what is called lub over IP which appears to be some sort of backhaul specification specific to Cellular / Wireless carrier technology. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOS Interface
Jorge Mendoza wrote: Brent Davidson wrote: Jorge Mendoza wrote: Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ You're comparing to apples to Orange. IOS is the Cisco operating system run by the Femtocells, not the protocol. I'm not familiar with Femtocells, but as far as I can tell (from reading wikipedia) they apparently can do SIP internally, but that is a more advance configuration and might require some additional software. Looks like they are more designed to what is called lub over IP which appears to be some sort of backhaul specification specific to Cellular / Wireless carrier technology. AFAIK, IOS interface for femtocell is a 3GPP2 specification. It is an application, no related to Cisco OS. Jorge Mendoza Interesting. I've never seen anything refer to IOS other than in the context of the OS run by Cisco routers although with so many acronyms around I suppose it's just a given that some of them should have more than one meaning depending on the context. Anyway, as I've already gone way past my level of understanding on the subject I'll leave this thread to someone more qualified to weigh in. :-P ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY
Cary Fitch wrote: It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8 bits each, and that is 2^140^8, a nearly inexhaustible key number which is related to audio and video data simultaneously stored on a Google Database, which is then sent to the user. Thus with the 140 byte message, full audio and video can be retrieved. This is an outgrowth of the data compression program circa about 1992, when disks were much smaller than today. A very small compression program would infinitely compress data on a disk to allow storage of more data. It was only a 200 bytes or so in size (DOS days):-) and worked perfectly. Running it once resulted in lots of storage space. It took very little time. Of course rewriting the MBR (Master Boot Record) takes very little time. Recovering the compressed data was tough though. Cary Fitch 04/01/09 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Wednesday, April 01, 2009 11:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY On Wed, Apr 01, 2009 at 06:52:55PM +0300, Dovid Bender wrote: I wish we could have this for real Micro-video-blogging: Limited to 140B ? I thought maybe it used Infinite Monkey Compression where a mathematic equation whose output over a specified domain would recreate the data-bits. For those unfamiliar with Infinite Monkey Compression it was theorized by me a few years ago as an offshoot of Infinite Monkey Theorem (monkeys, typewriters Shakespeare, etc...). The original theory was that is an infinite number of monkeys could eventually type the complete works of Shakespeare through random coincidence then a random bit generator running for an infinite amount of time would eventually produce the equivalent bit sequence of any particular piece of software. Infinity being, well, rather infinite and humans being mortal and all, infinite runs on a RBG didn't seem like all that great of an option, so I kept thinking... Then I realized that any file can be represented by a sequence of numbers. All you have to do is find the equation that will output those number sequences and you've got a highly-compressed way to recreate any file. Just send the equation give it a start and end value and let the computer save the output as a binary file. Unfortunately I was never able to take IMC beyond the purely theoretical. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording the calls
Both Montior and MixMonitor are part of the standard Asterisk distribution. There is no need to download anything else. bilal ghayyad wrote: Thsi Monitor CMD is a part of AsteriskNow so I have to download AsteriskNow and then take only the Monitor CMD or I can download the Monitor CMD alone? From where? Regards Bilal --- On Wed, 3/25/09, Steve Totaro stot...@first-notification.com wrote: From: Steve Totaro stot...@first-notification.com Subject: Re: [asterisk-users] Recording the calls To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, March 25, 2009, 8:57 AM On Wed, Mar 25, 2009 at 8:39 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I need to use the recording for the calls, did anyone try this on Asterisk? How it works? By the way: Asterisk support recording or it is another module that I have to download it and install it? Stable? Regards Bilal Applications Monitor or Mixmonitor should be fine. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast/Hyla/IAX Scalability?
David Backeberg wrote: On Sat, Mar 14, 2009 at 12:00 AM, Steve Underwood ste...@coppice.org wrote: Fully open-to-the-public FAX servers tend to get just get a lot of bad calls, many of them wrong numbers, or voice users. FAX servers for I've definitely seen that, and have been able to either identify the validity of a caller by CID or by calling the number and confirming a blast of fax tones. clue what kind of failure rate might be expected. You can find a bit more about these issues and our results at http://www.soft-switch.org/spandsp-soft-fax-performance.html After reading that, it occurred to me that I'm running SpanDSP 0.0.5 and 0.0.6 seems to have enhancements that may solve the problems I've been seeing. I'm convinced that it's worth upgrading and seeing if I can reduce my failure rate. Your differing failure rates between using ReceiveFAX and using iaxmodem seem to indicate your results relate to issues in your own system, I think I wasn't very good at setting it up, as I had no experience with IAX. Likely my fault rather than anything inherently wrong with the software. There were more moving parts than I was able to get a handle on, and when I switched to 1.6 and app_fax things 'just worked'. This is why I keep recommending the 1.6 approach over the 1.4 + IAX + IAXModem + Hylafax. LANs don't loose packets), will have a true failure rate (i.e. a rate of calls failing which had the potential to succeed) well below 1%. The That's consistent with my testing before I set it live. You mentioned recording faxes. I know how to do that with IAXModem, but are you familiar with a method for 1.6 and app_fax? I read through app_fax.c and didn't see any way to send a flag. Is the recording built into SpanDSP, or is is something IAXModem added on themselves? For what it's worth, the company I work for switched from WinFax to HylaFax last spring. We only have 4 analog phone lines coming in to a 4-port modem card, but the Hylafax system runs on the same server as our main Asterisk PBX. So far Hylafax is performing much better than WinFax ever did. When we have errors either sending or receiving, it is always either line problems or the wrong number being dialed resulting in a voice call to the fax line. I would estimate that our overall success rate is around 95% if you disregard faxes to wrong numbers or incoming voice calls to the fax lines. Load testing a large-scale fax system under real-world conditions is difficult if not impossible without having access to a variety of hardware and software fax devices scattered all over your prospective send or receive area. If you load test from your own location by attaching a bunch of fax machines or a fax sending server to your outgoing lines and have them dial back in, then you're only looping through your local telco's switching center. You might get very different results from sending faxes from out of state, or even across town. It's been my experience that telephone line quality varies greatly from place to place and even from time to time. A perfect example is from back in my days as a systems admin for a dial-up ISP. We were operating in a small town where PRI or channelized T1's weren't available so we had a bank of about 100 US Robotics external modems connected with serial cables to 2 Livingston PortMaster terminal servers. Everything would run fine (or as fine as it ever got with dial-up) until it decided to rain. Everytime we'd get more than a tenth of an inch of rain a large group of the modems would go haywire and start dropping calls. A couple of the modems would burn out completely. We had the telco out repeatedly and they always gave us some answer that didn't make any sense. After about the 6th time this happened they sent out a technician with a brand new line analyzer that happened to include a TDR. The vast majority of the lines we were having trouble with showed to have a partial short about 100 feet from our building which just happened to be right under the middle of the road in front of our building. They dug the section of line up and found that the cable had been partially cut at some point in the past and the wires were spliced with electrical tape and the whole bundle had then been wrapped with tape. Every time it rained, the water would seep into the shoddy splice and short all the lines together. When the water dried out, the shorts would go away and the lines would go back to normal. I've seen situation like that enough to know that until everybody has a purely digital phone line, there will always be line quality problems that will be out of the end user's control. Even though the company I work for now is a small company is a very rural area where technology is somewhat limited, we're beginning to realize just how antiquated Fax is becoming. E-mail and web services are rapidly replacing fax to the point that 90% of
Re: [asterisk-users] Help Inbound number
1246463 is not the same as 246463. Note the missing 1 If you want to match what is being dialed then your extensions.conf should look like this: [default] exten = 246463,1,Answer(SIP/8003) Bayardo Sanchez wrote: in my extension.conf i set : [default] exten = 1246463,1,Answer(SIP/8003) On Mon, Mar 16, 2009 at 12:06 PM, Pascal Bruno tipas...@gmail.com mailto:tipas...@gmail.com wrote: Do you have an extension set for 246463 in your extensions.conf? On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez bayardo.sanc...@gmail.com mailto:bayardo.sanc...@gmail.com wrote: i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com mailto:bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com mailto:bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd occurrence
Danny Nicholas wrote: Greetings listers, I am running Asterisk 1.4.21.2 on Suse 11.0 on a Dual Processor Dell Poweredge 1650. I recently attempted to update the BIOS and now have this happen: When the machine starts up, Asterisk runs fine. When I do a large wget or scp, the local SIP to SIP quality goes to heck in a handbasket. The only resolution I've found so far is to completely restart the machine. Obviously this is unacceptable. Has anyone else had this type of thing occur? Thanks in advance Danny Nicholas Are you using an on-board nic? If so, then it's possible the bios upgrade changed an operating parameter for the NIC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dictate
amit mehta wrote: Hello Members, Sorry for hijacking the earlier thread and asking the question last time. Is anyone aware about a solution to call incoming number and dictate the files by using Dictate feature of Asterisk used for Medical Transcription industry. Thanks Regards, Amit Mehta I'm not quite sure exactly what you're asking, so I'll cover what I see as answers to three possible scenarios. If all you want to do is read out the contents of text files, then look at the Cepstral text to speech engine. It would be fairly simple to build a script that parsed a list of files, read some form of numeric identifier to the user that allowed them to select a file to be read, then the file is sent to the Cepstral (app_swift) module and the contents of the file are read back to the user. You would probably need to implement some sort of pause, go back 1 sentence, go back one paragraph, etc controls as well. If you're looking for a way for a user to call in and record a voice file that will be later transcribed, that is probably easier than the reading a text file back. Just set up a macro that prompts for the callers ID, Patient ID, or whatever info you need using the Read function, then use the Record function to record the fileand save it with the previously gathered info in the filename (easiest solution) or store the recording and all the other info in a database (A bit more complicated). If you're looking for a way to allow a caller to read some information and have the system save that as a text file, then you'll need to talk to someone with more knowledge than me. Speech recognition isn't very easy right now. Best of luck, -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Read App Issues
Robert Broyles wrote: Okay. I'm using this all over SIP Trunking with Vitelity. Any other suggestions? -- Regards, Robert Broyles Eric Wieling, Asteria Solutions Group wrote: Robert Broyles wrote: So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has had this problem. When calling from an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in once. When testing the dialplan internally, it accepts only the digits that I key in. Anyone else experienced this? Yes. Most of the time it is either because I put relaxdtmf=yes in zapata.conf or because my rxgain is too low on that port. I've seen an issue similar to this when the sip peer was providing DTMF over multiple encodings at the same time. Usually, it's when Asterisk is expecting DTMF via inband, but the peer is sending inband and either INFO or rfc2833. What do you have the dtmfmode= line set to in your sip.conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Read App Issues
/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '5' received on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '5' on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '5' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:10] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '5' on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '6' received on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '6' on SIP/carrier-c4022740 [Feb 26 12:15:10] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '6' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:10] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '6' on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '6' received on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '6' on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '6' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:11] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '6' on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '7' received on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '7' on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '7' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:11] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '7' on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '7' received on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '7' on SIP/carrier-c4022740 [Feb 26 12:15:11] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '7' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:11] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '7' on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '8' received on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '8' on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '8' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:12] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '8' on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '8' received on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '8' on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '8' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:12] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '8' on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '8' received on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '8' on SIP/carrier-c4022740 [Feb 26 12:15:12] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '8' received on SIP/carrier-c4022740, duration 20 ms [Feb 26 12:15:12] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '8' on SIP/carrier-c4022740 [Feb 26 12:15:13] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '9' received on SIP/carrier-c4022740 [Feb 26 12:15:13] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '9' on SIP/carrier-c4022740 [Feb 26 12:15:13] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '9' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:13] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '9' on SIP/carrier-c4022740 [Feb 26 12:15:13] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '9' received on SIP/carrier-c4022740 [Feb 26 12:15:13] DTMF[3564]: channel.c:2284 __ast_read: DTMF begin ignored '9' on SIP/carrier-c4022740 [Feb 26 12:15:13] DTMF[3564]: channel.c:2212 __ast_read: DTMF end '9' received on SIP/carrier-c4022740, duration 100 ms [Feb 26 12:15:13] DTMF[3564]: channel.c:2265 __ast_read: DTMF end passthrough '9' on SIP/carrier-c4022740 After further testing, it seems to only be a problem when the same digit is entered 2 times or more in a roll. Any of the digits received with duration of 20ms aren't supposed to be there, but they show up anyway. Can someone else check this on their system, and see if this is a problem? -- Regards, Robert Broyles Brent Davidson wrote: Robert Broyles wrote: Okay. I'm using this all over SIP Trunking with Vitelity. Any other suggestions? -- Regards, Robert Broyles Eric Wieling, Asteria Solutions Group wrote: Robert Broyles wrote: So I'm using the READ
Re: [asterisk-users] Dropping RTP packets
You need canreinvite=no in the config for your sip phone and the veracity connection, otherwise Asterisk will just mediate the call setup then try to allow the sip phone and veracity to talk directly to one another. Jim Dickenson wrote: I have a SIP phone at home behind a NAT router registered with an * box at my office with a routable static IP address running version SVN-branch-1.6.0-r175638M. If I make a call from my SIP phone out a PRI circuit to my cell phone everything works as expected. I hear audio in both directions and all is good. If from the same SIP phone I make a call via our Veracity SIP account to my cell phone I hear no audio in either direction. In trying to find out what is wrong I used tcpdump to see if I could learn anything. I can see the phone sending fixed length UDP packets on to my home network heading to the IP address of the * box. If I run tcpdump on the * box I do not see the packets being received. I do not see the * box sending any packets to my home network either. I have not checked if the * box is receiving packets from Veracity I only know that no audio packets are sent to my home network. If I use tcpdump to watch the SIP phone call via the PRI circuit I see packets both on my home network and my * box. If I use a SIP phone located in my office and make a call via Veracity everything is okay. Also a co-worker has a vpn router on his home network connected to the office vpn server and he can make calls from his SIP phone via Veracity without problems. I can also call his SIP phone from my SIP phone and packets pass as expected. It seems as if audio packets from my SIP phone disappear only if they are involved with a call via Veracity. Does anyone have some idea what I might look at to find what is causing this problem? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SheevaPlug Development Kit
Yes please let us know how it works out. I have several projects in the works that this might work for. David fire wrote: please keep us informed about it. David 2009/2/25 Kristian Kielhofner kristian.kielhof...@gmail.com mailto:kristian.kielhof...@gmail.com Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk
If you are connecting 2 Asterisk boxes and you are setting them both up then use IAX (Intra Asterisk Exchange) instead of SIP. IAX does not pass off the packets to RTP and should fix some of the firewall problems people get. As far as credential passing I am unsure if it will help that. So like mentioned below set up a connection in iax.conf on both ends. There is a chapter in the O'Reily Asterisk the future of telephony book that talks you through an Asterisk to Asterisk connection using IAX. Is there a specific reason you want to use SIP/RTP? Brent Imanol Pardavila wrote: I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using a sip account (Asterisk 1 acting as a conventional sip user). Thanks Regards Danny Nicholas escribió: Inter-* registry is done with iax.conf, not sip.conf. sip is for phones/sip-lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Imanol Pardavila Sent: Thursday, January 29, 2009 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] howto configure an asterisk to send credentials in a REGISTER message to another asterisk Hi, I am trying to register an asterisk (Asterisk 1) against another one (Asterisk 2). My problem is that the REGISTER message goes without credentials and the Asterisk 2 send a 401 message to the Asterisk 1. How can I configure Asterisk 1 to force it to send credentials? I have tried setting Asterisk 2's IP in the realm field of Asterisk's 1 sip.conf, but it doesn`t work. Any ideas? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for SIP loud ringer
If you know anyone with electronic experience you could take the speaker output from a SIP phone either by connecting to the internal wires or if there is a headset jack to there. Run that speaker output into an amplifier to some speakers. This would amplify all speaker output so going to speaker phone on that phone would not work but you said the room was noisy so going speaker phone would not work anyway. So where do I get an amplifier speaker system? Your local computer store with a set of standard computer speakers. I have a Grandstream Budgetone 200 that has a headset jack on the back. I am not sure if ringing comes out the ear piece on a head set but if it does just make an adapter from the 1/8 speaker jack to the smaller one on the back of the phone, plug in the computer speakers and see what happens. It would require an extra phone in that room on the same ring group as the other line but it should work and you could make it a peer ( I think I got that right incoming calls only ) so no one could call out on it. Brent Mike wrote: Danny, Thanks for the idea, I thought of it but I was looking for a more elegant solution, and one that would as much as possible not require my intervention in any way. A PC requires support even in the best of times: it`s got harddrives, software, patches, etc, etc… An alternative would be a SIP phone with a very loud max ring, but that`s not the case with the phones I know (Polycoms) Mike *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, January 28, 2009 10:45 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Looking for SIP loud ringer Why don’t you put a PC in the storeroom with a softphone to be the “loud ringer”? You could make the ring though the speakers be as loud as the system would support. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mike *Sent:* Wednesday, January 28, 2009 9:36 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Looking for SIP loud ringer Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing DTMF
I have had this same problem using via:talk. Even though they tell me they have hundreds of people using Asterisk with their service that have no problems we cannot make it work. I have also had reponses confirming that in this email list. So don't wast your time with via:talk. Christopher Gray wrote: Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the question: does anybody know of a carrier that can reliably allow an extension in my pbx to send touchtone to a calling party? I have tried Vitelity and VoicePulse. Neither can do this, and VoicePulse indicates they know it's a problem and will fix it at some unknown time in the future. For the curious, here is the reason for the need. My wife, who works as a translator, will use this extension to receive calls from companies needing translation. When she receives such a call, step 1 for her is to enter an employee id code. At the end of the call, she must enter an additional code to receive an ending time. Vitelity can't do this at all. VoicePulse works about 75% of the time which is not acceptable. Thanks for any advice. Chris Christopher Gray, President Bay Area Digital Promoting good health with innovative technology 870 Market Street, #653 San Francisco, CA 94102 Phone: (415) 217-6667 fax:(415) 962-2520 Email: ch...@bayareadigital.us ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dead sip channel
Jerry Geis wrote: hi, try to set the rtptimeout value in sip.conf to a resonable value - so asterisk will kill the channels if it does not receive rtp traffic for the specified time regards, Wolfgang I uncommeted the rtptimeout=60 value in sip.conf and did a reload. It still hasnt dropped the dead call after a couple minutes now... Do I have to stop and start again? Was hoping it would just drop the call and continue on. Jerry Sounds like the problem is that the slow computer is no longer accepting calls after the first. Is Asterisk running on that machine as well? If so, check to see what it says about the sip channels. If not, you will need to look into the software running on that machine and try to figure out why it is either not hanging up or why it is dieing after it gets a call. -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting observation
I investigated Charter for our business phone systems and asked many of these questions of the sales person. I was told they have a dedicated part of the bandwidth available that is used just for phone traffic. I could break out my college networking book and get you the frequency break down as far as what is used for IP and what is used for TV and why the upload and down load speeds are asymmetrical if I was motivated but I am not so you will have to take this for what it is worth. As cable is not a point to point system (cable is shared bandwidth for all users on that cable) that means all phone users will be using the same piece of spectrum on that cable. This means that too many phone calls on that line at the same time could affect a Charter phone call. I do not know if they use analog or digital signals for the phones but if we use the cell phone system as an example they took down all analog towers because they could service more phones on the same bandwidth with digital. I would assume that would hold true for the spectrum on a cable as well. I would also find it hard to believe that they would not use off the shelf technology. That being said my brothers in-laws are using it and are having no problems what so ever. David Gibbons wrote: snip My understanding is that Charter 'telephone' doesn't use IP at all but rather uses some additional frequency spectrum on their cable network. Hence, the reason why faxing with their service is reliable unlike other providers who are *actually* using VoIP. /snip I think what you're referring to is the general hesitance of the cable providers to call their phone service VOIP service. VOIP still has a negative connotation with most regular folks, so they don't want to negative PR. I'm don't have any facts, but I'll bet you a penny that they don't have a proprietary system using something /OTHER/ than IP to send encapsulated voice over 'additional frequency spectrum'. That would be prohibitively expensive to develop and pointless from a technical standpoint, given that IP telephony is already set to deploy and relatively mature. The reliability of faxing is based soley on network jitter and latency and codec compression. I've found that taking the compression out of the mix (using g.711 ulaw) and controlling the jitter and latency (something that's easy to do on a private network like theirs with QOS) causes faxing to be pretty darn reliable. --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Stealing
Look int the ChannelRedirect command. Geoff Lane wrote: Hi All, I'd appreciate some help on how to implement call stealing. That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on my living room extension and need to refer to some paperwork, I can go to the study, pick up that extension, dial *46, and the call is transferred to the study where I can continue the call with the paperwork to hand. It also helps if you take a call for someone else if that person can steal the call from your extension. Call parking provides a partial work-around but it's a pain having to remember to park a call before moving location. I haven't found an application for call stealing and can't figure out a way to do this. Can anyone help? TIA, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL question: testing channel variables
Initialize FOOBAR to some know value (ie NO) and change it when you need
to. Then it will always be defined.
Klaus Darilion wrote:
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=YES
Of course I could use the following code, but this bloats up the code:
if (${EXISTS(${FOOBAR})}) {
if (${FOOBAR}=YES) {
...
}
}
Is there another syntax to have nice looking code but avoid the warning?
thanks
klaus
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Re: [asterisk-users] Not Dialing 9
Thczv F. Thczv wrote: When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If I dial 1, then anything other than 0 or 1, Asterisk knows I am dialing long distance. If I start with any number other than 1, Asterisk knows I am dialing a local or local toll call. This has worked fine for me (as far as I know). Is there some flaw I am not seeing? I see a lot of small businesses that require a 9 to dial out, even though they don't have very many extensions. Couldn't they do what I did and not have to dial 9? I ask because we are having a problem where I work with our Cisco 7940 phones adding an extra 1 sometimes, which gets the local Sheriff upset (too many 911 calls). Thanks, Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I think you are over thinking this. We set our Asterisk server up with multiple outgoing dial rules to handle local and long distance. Keep in mind that we are connected to SIP provider that takes care of some of this for us. Our local extensions are 2** I have outgoing call rules similar to exten = _NXX!,1,Dial(outgoing sip connection) ( this is for local calls ) exten = _NXXNXX!,1,Dial(outgoing sip connection) ( this is for long distance calls ) exten = _911!,1,Dial(outgoing sip connection) (this is of course for dialing 911 ) exten = _011XX.,1,Dail(yada yada yada) (interntional calls) exten = _1NXXNXX!,1,Dial(Yada yada yada ) (long distance with a 1 in front.) If you have to have a 1 on front of your long dist numbers ( we don't ) leave off the _NXXNXX! pattern and only use the last one. Asterisk will now look at the relevant extensions and decide which to use. Using this method if you mis-dial then the phone line does not get used and Asterisk sends a failed number sound. If you dial 3 numbers it will look at the local extension you have set up and if the one you dialed exist it dials it. If you get an error. If you match the patern above it dials out. The only time with this system you would have a problem is if you make your internal extensions 7 or 9 digits. Doing like I have above Asterisk will match your pattern more based on ext length than on what order the numbers are in. Hope that helps -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Watkins, Bradley wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent
Davidson
Sent: Wednesday, December 31, 2008 1:03 PM
To: m...@digium.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] AEL Variable Warning Messages
Well, before I file a bug I have another question... In AEL,
what is the correct syntax? Do all variable references still need to be
wrapped in ${} or not? If they do, then the documentation on
voip-info.org needs to be changed to reflect that.
Yes, variable references need to be wrapped in ${}. Where on the wiki
do you see an example that is otherwise? I just looked at the main
documentation for AEL, and I didn't see any instances of it. Certainly
they can and should be fixed if they are there.
Beyond that, what are the rules for putting the values assigned
to variables in quotes? In my example above, at one point I had a space
between the = and the Zap/r2 statement with no quotes. The value
assigned to TRUNK then included a leading space. I didn't test to see
whether or not putting a space after the variable name adds a space to
the variables name. I would think that any spaces after an operator
should be ignored unless the come after a single or double quote.
The rules are that there aren't any really. Neither a single- nor
double-quote have any specific meaning in the sense of signifying a
string. I'm also curious to know where you saw an example of assignment
that used quotes of any kind, since I can't find that either.
Regards,
- Brad
Both the AEL and AEL2 examples include the following examples in the
Variables section:
---HTML Copied from WIKI page
http://www.voip-info.org/wiki/view/Asterisk+AEL---
Variables
Variables in Asterisk do not have a type, so to define a variable, it
just has to be specified with a value.
Global variables are set in their own block.
globals {
CONSOLE=Console/dsp;
TRUNK=Zap/g2;
};
*NOTE:* The opening curly-brace must appear as above. Moving it to the
following line may have disastrous consequences!
Variables can be set within extensions as well.
context foo {
555 = {
x=5;
y=blah;
divexample=10/2
NoOp(x is ${x} and y is ${y} !);
};
};
*NOTE:* Asterisk beta1 parses assignments using a $[] wrapper as opposed
to the more logical way of doing it like Set and SetVar work. In this
example, I had ${ARG1} set to SIP/x7065558529 sans-quotes and it
flunked out.
*NOTE:* Another opinion: The $[ ] allow expressions to be used, and add
extra power to the language. Read the README.variables about the
requirements of $ http://www.voip-info.org/wiki/view/Asterisk+AEL
expressions. In the following example, the SIP/x7065558529
should not be sans quotes. So, the statement might have been entered: requesting_channel=${ARG1};( where the 's prevent the evaluation. )
*NOTE:* These things are wrapped up in a $[ ] expression: The while()
test; the if() test; the middle expression in the for( x; y; z)
statement (the y expression); Assignments --- the right hand side, so a
= b - Set(a=$[b])
requesting_channel=${ARG1}
ERROR: Oct 10 12:48:59 WARNING[19726]: ast_expr2.y:811 op_div:
non-numeric argument
--- Executing Set(SIP/x7065558529-2afd, requesting_channel=0) in new stack
FROM show dialplan:
's' =1. Set(requesting_channel=$[ ${ARG1} ]) [pbx_ael]
But you can use Set and it works the old way.
Set(requesting_channel=${ARG1})
Writing to a dialplan function is treated the same as writing to a
variable.
context blah {
s = {
CALLERID(name)=ChickenMan;
NoOp(My name is ${CALLERID(name)} !);
};
};
---END HTML Copied from WIKI---
I see the line about using the old Set syntax, but the examples and the
rest of the notes imply that that you could just do x=5; rather than
${x}=5; and get the same result either way.
Another question along these lines... If I set a Global called TRUNK
in the globals section and later assign do a TRUNK=whatever it appears
that a local variable called TRUNK is created instead of using the
global. You must explicitly use the Set(GLOBAL(TRUNK)=whatever) syntax
to alter the global.
As far as the quotes go, I resorted to using them in an effort to get
rid of that extra space at the beginning of my value. I just figured
that if the examples were wrong on one aspect that maybe they were off
in other ways as well.
The examples in the Macros and Conditionals sections of AEL shows
the variables with the curly brace syntax, but the loops section uses
both syntaxes in it's examples.
I would be happy to help clean up the examples on the wiki as long as
I'm sure I know what I'm doing. As for my dialplan, I've reverted to
using the Set
Re: [asterisk-users] AEL Variable Warning Messages
Benoit wrote: Brent Davidson a écrit : Another question along these lines... If I set a Global called TRUNK in the globals section and later assign do a TRUNK=whatever it appears that a local variable called TRUNK is created instead of using the global. You must explicitly use the Set(GLOBAL(TRUNK)=whatever) syntax to alter the global. And the question is ? I guess my question was whether or not the above is the intended behavior. It seems to me that if you declare a global, then later use the same name it should refer to the global without having to use the old syntax. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Steve Murphy wrote:
On Tue, 2008-12-23 at 12:14 -0600, Brent Davidson wrote:
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
defined 2 channel groups in my zapata.conf. The second office needs all
of their outgoing calls to go out over their lines so the people they
call will have the correct callerID. I created an asterisk database and
with entries in the database for all extensions in the second office and
defined the following macro:
globals {
CONSOLE=Console/dsp;
TRUNK=Zap/r1;
TCTC_Operator=15;
Law_Operator=12;
};
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
It's working and correctly routing outside calls, but I get the
following messages when I reload the extensions.ael file:
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
Any idea what is causing the warnings?
Yes, I do! I was concerned that users were falling into a common
error, where they forget to wrap variable references in $(); so,
if it looks like an expr has arithmetic operators, but no variable
refs, then you get this message.
Yes, I *could* have made it more intelligent. File a bug, and I'll
see if I can do so. At the worst, you can ignore this warning, or
I can simply remove this overly-simple warning.
murf
Well, before I file a bug I have another question... In AEL, what is
the correct syntax? Do all variable references still need to be wrapped
in ${} or not? If they do, then the documentation on voip-info.org
needs to be changed to reflect that.
Beyond that, what are the rules for putting the values assigned to
variables in quotes? In my example above, at one point I had a space
between the = and the Zap/r2 statement with no quotes. The value
assigned to TRUNK then included a leading space. I didn't test to see
whether or not putting a space after the variable name adds a space to
the variables name. I would think that any spaces after an operator
should be ignored unless the come after a single or double quote.
Thanks,
Brent
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[asterisk-users] DTMF does not work
I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on the one and now we don't drop any longer. doh!!! Last, We are having DTMF problems with our provider (via:talk). Does anyone have any experience with them and if so can you share it? via:talk does have a sample sip.conf and extensions.conf file to use but the dial plan they set up does not require any DTMF so they may never have tested it. We have tried inband, auto, rfc2833 for our DTMF and nothing works. I have submitted a ticket with them but the last time I did that they never responded so that is why I am posting here. I signed up with another SIP provider for a test account and the DTMF passes no problem from them so I must conclude there is some setting that via:talk has that is causing the problem. via:talk will not confirm this but they must be using Asterisk as all the menus and such they have feel very Asteriskish. Is there something I can tell via:talk to try on their end to make this work? As a side symptem every time our system registers with via:talk it seams to jump from server to server on their end. They must have some sort of load balancing going on that is causing that. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?
Kristian Kielhofner wrote: On Tue, Dec 23, 2008 at 4:40 AM, Steve Totaro stot...@first-notification.com wrote: It's all ball bearings these days What is the deal with Fletch quotes these days? Don't get me wrong, I appreciate them but I'm starting to wonder where this is all coming from. I *think* it's because Fletch has been on HBO lately. Am I correct? Who doesn't like a good fletch quote. Thank you very little ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Problems
First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on the one and now we don't drop any longer. doh!!! Last, We are having DTMF problems with our provider (via:talk). Does anyone have any experience with them and if so can you share it? via:talk does have a sample sip.conf and extensions.conf file to use but the dial plan they set up does not require any DTMF so they may never have tested it. We have tried inband, auto, rfc2833 for our DTMF and nothing works. I have submitted a ticket with them but the last time I did that they never responded so that is why I am posting here. I signed up with another SIP provider for a test account and the DTMF passes no problem from them so I must conclude there is some setting that via:talk has that is causing the problem. via:talk will not confirm this but they must be using Asterisk as all the menus and such they have feel very Asteriskish. Is there something I can tell via:talk to try on their end to make this work? As a side symptem every time our system registers with via:talk it seams to jump from server to server on their end. They must have some sort of load balancing going on that is causing that. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pattern Matching
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is present, replace it with Unknown. I'm using the
ael format for my dialplan and have been looking for a way to do this,
but haven't found anything yet. Is there a way to do this inside the
dialplan or do I have to pass it out to an AGI script?
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[asterisk-users] AEL Variable Warning Messages
I have two offices sharing a phone system. They also share a common
internal context because all of the employees of the second office also
work for the first office. Each office has 4 outside lines and I have
defined 2 channel groups in my zapata.conf. The second office needs all
of their outgoing calls to go out over their lines so the people they
call will have the correct callerID. I created an asterisk database and
with entries in the database for all extensions in the second office and
defined the following macro:
globals {
CONSOLE=Console/dsp;
TRUNK=Zap/r1;
TCTC_Operator=15;
Law_Operator=12;
};
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
It's working and correctly routing outside calls, but I get the
following messages when I reload the extensions.ael file:
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
Any idea what is causing the warnings?
Thanks,
Brent
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Re: [asterisk-users] Pattern Matching
Philipp Kempgen wrote:
Brent Davidson schrieb:
On my asterisk system, if an incoming call only has a number for the
caller ID and no name, the system is using the channel name as in the
Callerid Name field. I would like to use some sort of pattern match
test to test for the presence of Zap/ in the ${CALLERID(name)}
variable and if it is present, replace it with Unknown. I'm using the
ael format for my dialplan and have been looking for a way to do this,
but haven't found anything yet. Is there a way to do this inside the
dialplan
if (${CALLERID(name):0:4} = Zap/) {
Set(CALLERID(name)=Unknown);
}
Not sure why you would want to put the channel name into the
caller ID name in the first place.
Philipp Kempgen
Thanks all. As far as why the channel name is in the caller ID, I don't
know. I'm certainly not doing it intentionally. I don't have any code
in the dialplan that even touches the CallerID, so I guess Asterisk is
doing somehow when the Name part of the CallerID is unknown... Either
that or my Snom 300 phones are picking the wrong info to use for CallerID.
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Re: [asterisk-users] AEL Variable Warning Messages
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);
Philipp Kempgen
According to the AEL Documentation I should be able to set variables
without using the Set command. They even give the following example:
context foo {
555 = {
x=5;
y=blah;
divexample=10/2
NoOp(x is ${x} and y is ${y} !);
};
};
I wonder if maybe AEL is ignoring the double quotes and treating the
Zap/r2 as if it were division??? Should I file a bug report on this?
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Re: [asterisk-users] AEL Variable Warning Messages
Dave Fullerton wrote: I had gotten similar messages when I forgot to put quotes around channels like that (took me forever to realize that one). Since you have them I would say this is a bug. What version of asterisk are you running? -Dave I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with DAHDI and Rhino RCBFX card. I tried doing a new install with 1.4.22 yesterday and couldn't get Oslec to work correctly with the Rhino card when running with DAHDI instead of zaptel. Unfortunately 1.4.22 no longer has Zaptel. :( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Tzafrir Cohen wrote: On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote: Unfortunately 1.4.22 no longer has Zaptel. :( Asterisk 1.4.22 builds with both Zaptel and DAHDI. I spent several hours trying to make it work yesterday and it just wouldn't. I kept getting an error message that it was unable to bind the echo canceler to channel 1. It might have something to do with the RCBFX drivers, I'm not sure. I found your page and followed your instructions. Everything appeared to work until I checked with dahdi_cfg -vv. That's where I got the message. Don't have my notes here so I don't have the actual error message right now. -- Brent Davidson I.T. Manager Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Jeff LaCoursiere wrote: On Tue, 23 Dec 2008, Brent Davidson wrote: Dave Fullerton wrote: I had gotten similar messages when I forgot to put quotes around channels like that (took me forever to realize that one). Since you have them I would say this is a bug. What version of asterisk are you running? -Dave I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with DAHDI and Rhino RCBFX card. I tried doing a new install with 1.4.22 yesterday and couldn't get Oslec to work correctly with the Rhino card when running with DAHDI instead of zaptel. Unfortunately 1.4.22 no longer has Zaptel. :( Why do you need oslec to work with the rhino card - it has hardware echo cancellation built in doesn't it? j The Rhino card is supposed to have hardware echo cancellation. That's one of the main reasons I switched to that card from the X-100p's I was using. Unfortunately, either I don't know how to turn on the hardware echo cancellation or it just doesn't work. I have 5 separate location where I'm using that card and if I turn off Oslec at any of them the echo is so bad that the systems is virtually unusable. With Oslec enabled, however, there is no echo at all. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Variable Warning Messages
Tzafrir Cohen wrote: What error message from where? With Zaptel the echo canceller settings are global (that is: one hard-coded echo canceller). With DAHDI there are echo canceller modules and you can (and actually need to) set them per-channel. It might have something to do with the RCBFX drivers, I'm not sure. I found your page and followed your instructions. Everything appeared to work until I checked with dahdi_cfg -vv. That's where I got the message. Don't have my notes here so I don't have the actual error message right now. I don't remember the actual error name, but it showed up when I did dahdi_cfg -vv. It was something like DAHDI_ATTACH_ECHO_CANCELLER Failed for Channel 1. Unsupported command (22). I was trying to see if maybe it was logged to my syslog but this is all I find in my /var/log/messages: Dec 22 17:01:43 localhost modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/2.6.23.8x86_64/dahdi/dahdi_echocan_osle c.ko): Unknown symbol in module, or unknown parameter (see dmesg) Dec 22 17:01:43 localhost kernel: rcbfx 1: Spotted a Rhino: Rhino RCB4FXO (4 channels) Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol oslec_create Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol oslec_update Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol oslec_free Also, when using the Dahdi/Oslec/RCBFX combination I was getting tons of blocks like this in my syslog: Dec 22 16:54:58 localhost kernel: 80c = 2c7e5000 Dec 22 16:54:58 localhost kernel: 810 = 240 Dec 22 16:54:59 localhost kernel: 814 = 0 Dec 22 16:55:00 localhost kernel: 818 = 0 Dec 22 16:55:00 localhost kernel: 81c = 0 Dec 22 16:55:00 localhost kernel: 820 = Dec 22 16:55:01 localhost kernel: 824 = Dec 22 16:55:01 localhost kernel: 828 = Dec 22 16:55:02 localhost kernel: 82c = 0 Dec 22 16:55:02 localhost kernel: 830 = Dec 22 16:55:02 localhost kernel: 834 = Dec 22 16:55:02 localhost kernel: 838 = Dec 22 16:55:03 localhost kernel: 83c = 0 Dec 22 16:55:03 localhost kernel: 840 = 3 Dec 22 16:55:04 localhost kernel: 844 = f Dec 22 16:55:04 localhost kernel: 848 = Dec 22 16:55:04 localhost kernel: 84c = 0 Dec 22 16:55:04 localhost kernel: 850 = 0 Dec 22 16:55:05 localhost kernel: 854 = 10f Dec 22 16:55:05 localhost kernel: 858 = 14e00ff Dec 22 16:55:12 localhost kernel: 85c = 3d434310 Dec 22 16:55:13 localhost kernel: 860 = 0 Dec 22 16:55:13 localhost kernel: 864 = 0 Dec 22 16:55:18 localhost kernel: 868 = 229e229e Dec 22 16:55:18 localhost kernel: 86c = 0 Dec 22 16:55:19 localhost kernel: 870 = 5 Dec 22 16:55:19 localhost kernel: 874 = 5 Dec 22 16:55:20 localhost kernel: 878 = Dec 22 16:55:20 localhost kernel: 87c = 0 Dec 22 16:55:21 localhost kernel: 880 = 0 Dec 22 16:55:21 localhost kernel: 884 = 0 Dec 22 16:55:21 localhost kernel: 888 = 0 Dec 22 16:55:22 localhost kernel: 88c = 0 Dec 22 16:55:22 localhost kernel: 890 = 0 Dec 22 16:55:22 localhost kernel: 894 = 0 Dec 22 16:55:24 localhost kernel: 898 = 0 Dec 22 16:55:26 localhost kernel: 89c = 0 Dec 22 16:55:28 localhost kernel: 8a0 = 0 Dec 22 16:55:28 localhost kernel: 8a4 = 0 Dec 22 16:55:28 localhost kernel: 8a8 = 0 Dec 22 16:55:28 localhost kernel: 8ac = 0 Dec 22 16:55:28 localhost kernel: 8b0 = 0 Dec 22 16:55:28 localhost kernel: 8b4 = 0 Dec 22 16:55:28 localhost kernel: 8b8 = 0 Dec 22 16:55:28 localhost kernel: 8bc = 0 Dec 22 16:55:28 localhost kernel: 8c0 = 0 Dec 22 16:55:28 localhost kernel: 8c4 = 0 Dec 22 16:55:28 localhost kernel: 8c8 = 0 Dec 22 16:55:28 localhost kernel: 8cc = 0 Dec 22 16:55:28 localhost kernel: 8d0 = 0 Dec 22 16:55:28 localhost kernel: 8d4 = 0 Dec 22 16:55:28 localhost kernel: 8d8 = 0 Dec 22 16:55:28 localhost kernel: 8dc = 0 Dec 22 16:55:28 localhost kernel: 8e0 = 0 Dec 22 16:55:28 localhost kernel: 8e4 = 0 Dec 22 16:55:28 localhost kernel: 8e8 = 0 Dec 22 16:55:28 localhost kernel: 8ec = 0 Dec 22 16:55:29 localhost kernel: 8f0 = 0 Dec 22 16:55:29 localhost kernel: 8f4 = 0 Dec 22 16:55:29 localhost kernel: 8f8 = 0 Dec 22 16:55:29 localhost kernel: 8fc = 0 Dec 22 16:55:29 localhost kernel: 900 = 0 Dec 22 16:55:29 localhost kernel: 904 = 0 Dec 22 16:55:29 localhost kernel: 908 = 0 Dec 22 16:55:29 localhost kernel: 90c = f0f0f0f Dec 22 16:55:29 localhost kernel: 910 = f0f0f0f Switching back to Zaptel solved all of the problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / TDM400P card stopped working
Tilghman Lesher wrote: On Monday 15 December 2008 00:57:08 Langdon Stevenson wrote: Hi Paul Thanks for the reply. I have removed and re-installed all of the Fedora Zaptel packages with Yum. I have the following installed: asterisk-zaptel 1.4.12.1-1.fc8 zaptel.i386 1.4.12.1-1.fc8 zaptel-devel.i386 1.4.12.1-1.fc8 zaptel-lib.i386 1.4.12.1-1.fc8 zaptel-utils.i386 1.4.12.1-1.fc8 The command: modprobe wctdm produces: FATAL: Module wctdm not found. This probably means that the modules were compiled for a kernel other than the one you have installed. You probably have multiple directories within /lib/modules, and the zaptel modules are in a directory other than what is listed with 'uname -r'. In this case, compiling from source is probably your best bet. This may be an obvious thing, but you didn't mention checking whether or not the card was still seated in the slot properly after the move. I know from experience that when you move offices, even if you take all the precautions possible, a card can get bumped just enough to jostle the connections loose. Even if the card appears to be seated correctly I'd take it out and re-seat it. Unfortunately it looks like you may have compounded the problem by removing and reinstalling the zaptel packages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP CallerID Question
I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says Office1. The same thing happens if a person at Office1 transfers an incoming call to Office2. The caller ID at Office2 always just says Office1. What I would like to happen would be when Bob at Extension 12 at Office1 calls Office2 the caller ID at office 2 would say Bob in the name files and 12 in the number field. If Bob does a blind transfer to an extension at Office2 I would like the caller ID on the Office2 phone to display the original caller's name and number. I've read most of the documentation on the CallerID variables, but am still having a bit of trouble wrapping my head around the necessary logic to accomplish what I need to do, (mainly because I'm in the middle of a totally unrelated project and am having trouble multi-tasking). Could anyone give me a starting point? Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each office
has it's own server and prefix by which it is accessed from another
office. So for office1 to dial extension 12 at office2 he would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:[EMAIL PROTECTED],,Tt);
Hangup;
}
And in my SIP.conf on Office 1
[Office2]
username=Office1-user
fromuser=Office1-user
host=XXX.XXX.XXX.XXX (edited out)
type=peer
context=internal
secret= password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
qualify=yes
canreinvite=no
In My Sip.Conf on Office2:
[Office1-user]
username=Office1
host=XXX.XXX.XXX.XXX (edited out)
type=user
context=internal
secret=password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
canreinvite=no
Separating into peer and user entries was the only way I was able to get
calls to go through and be authenticated properly. Would this setup
have any bearing on the caller ID?
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Re: [asterisk-users] SIP CallerID Question
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each office
has it's own server and prefix by which it is accessed from another
office. So for office1 to dial extension 12 at office2 he would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:2...@office2,,Tt);
Hangup;
}
I don't see anything sticking out as being wrong. For kicks, what is the
output of sip show user Office1-user on office2?
___
localhost*CLI sip show user Office1-user
localhost*CLI
* Name : Office1-user
Secret : Set
MD5Secret: Not set
Context : internal
Language : en
AMA flags: Unknown
Transfer mode: open
MaxCallBR: 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 20
Callgroup:
Pickupgroup :
Callerid :
ACL : No
Codec Order : (speex:20)
Auto-Framing: No
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Re: [asterisk-users] SIP Registry Problems
Stefan I tried this and now I get this: -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host grimlock.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host grimlock.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.optimusprime.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.optimusprime.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host grimlock.vtnoc.net, port 5060 The connections are all over the place and I still don't get DTMF to pass. Any other suggestions? Stefan Schmidt wrote: Brent Vrieze schrieb: Here is what happens: 1. Asterisk verifies connection to the server and we get this. (CLI output) -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 It jumps around from server to server all the time. hello, you should add the second and third server to your user.conf as a friend too, they just use something like a load balancer but you only accept calls from one of their 3 servers. so i think what happens is that if a call comes from one of the server you didnt authorize it just get an error like authorization required and then fallback to your boss cell number. maybe you could trace this with sip debug. best regards. steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Registry Problems
Having big problems and for months. Our service provider (via:talk) says they are Asterisk friendly but they are not. Here are the specifics (please read the bottom of the msg too) System: Dell SM Business server 2GB RAM, Core II Processor (should be plenty) OS: open SUSE 11 Asterisk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a couple tweaks in users.conf that via:talk wants. Here is what happens: 1. Asterisk verifies connection to the server and we get this. (CLI output) -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 It jumps around from server to server all the time. 2. With all the server jumping sometimes incoming calls get re-routed by via:talk to the bosses cell phone, the fail safe dump off number. Seconds after calling and getting re-routed to the boss I call and it goes through. 3. We cannot recieve DTMF from via;talk, have tried auto, rfc2833, and inband without success with any of them, and yes we had via:talk change their end too. Here is the users.conf entry or the connection to via:talk. [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = phone number secret = blablabla trunkname = via:talk ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = phone number authuser = phone number insecure = port,invite dtmf = inband dtmfmode = inband relaxdtmf = yes ;rfc2833compensate = yes port = 5060 canreinvite = no disallow = all allow = ulaw,gsm I did set up a very basic Asterisk box yesterday that put all the conection settings in sip.conf and I even renamed users.conf so it could not load. I then put in about a 10 line hand coded dial plan in extensions.conf and got the same results Of course via:talk is of no help as they only officaly support the Linksys PAP2 they sent us with our account. My solution is to move away form via:talk and leave the problem behind. I then figured smeer nasty things on the internet about them but I'm too late, many other people already have. :) The problem with moving is we paid for a years service and that is up in April and the boss is cheap, cheap, cheap. Please help my connection woes and thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
John Todd wrote: Erik - Have you found RealSpeak to be worth the cost? Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? It's been a while since I did a head-to-head comparison between Cepstral and (anything else) so I did a quick demo of the RealSpeak Host-based telecom app: http://www.nuance.com/realspeak/demo/ (contact data required) and the Cepstral demo: http://www.cepstral.com/demos/ I used the Jill (default - 8khz) for RealSpeak and Allison (default) for the tests, and played back the same phrase: Congratulations. You have successfully installed and executed the Asterisk open source PBX. My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director This may not be a perfectly fair comparison. Looks like you're comparing the RealSpeak 8khz voice to the Cepstral default Allison which is NOT 8khz. If you look on the Cepstral site you'll see Desktop Voices and Telephony Voices. The Cepstral Telephony voices are 8khz, and I suspect their quality is on par with RealSpeak. I recently licensed the Allison-8Khz voice for some of the admin functions on my companies phone systems where I didn't want to record prompts and Flite was too robotic sounding. The Allison-8khz voice is virtually indistinguishable from the pre-recorded Allison prompts, except for maybe some minor differences in inflection. I was thoroughly impressed with the quality though. For the most part it sounds like you've hired Allison to record custom prompts. The Allison Desktop voice is OK, but sounds sort of like Allison is taking through a spinning fan blade. When you're doing TTS comparisons be sure you're comparing apples to apples and not peaches to apricots. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
