Re: [Asterisk-Users] Call Queue Transfer
On 04/29/06 10:06 Josué Conti said the following: is that if the agent transfers the call, for another user and this user takes care of the call, the status of the agent in the show agents is of that it the same continues speaking (talking to zap) with circuit how are you performing the transfer ? are they blind/attended transfers using the keystrokes in features.conf ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a way to monitor the DTMF tones on a channel?
Is there a way to monitor the DTMF tones on a channel? I have a prepaid application working in asterisk. When the user dials a call and wants to cancel the call before it is answered, there is now way to do it without hanging up and redialling the access number. Is there way to monitor a sequence of DTMF tones and cancel the call? If I use a SIP gateway or proxy rather than dial asterisk directly will that be possible? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stupid trick of the day (fried polycom)
[EMAIL PROTECTED] wrote: I've been playing around with a new system I'm going to install in another office. In setting up the Polycom's, I accidently used a new power supply from a new 601 (24VDC) with an 600. The 600 only require 12VDC. Now, I get nothing on the screen of the 600 when I plug in 12 VDC. (At the time, I didn't even realize the power supplies were supplying different voltages.) Yes, this is one of my peeves with the 601... they changed power supply voltages without changing connector styles, leading to this being a very easy mistake to make. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Mediatrix 1204
Hi all, Please excuse my newbie status I need help in configuring a mediatrix 1204 PSTN gateway with asterisk. Basically each FXO port is configured with a SIP username and automatic transfer extension, which should transfer incoming calls to an asterisk extension. I created extensions corresponding to the FXO port SIP usernames. Port 1 - SIP username - 21383396 - call forward to - 300 I am pasting 3 SIP messages between the Mediatrix (192.168.0.27) and Asterisk (192.168.0.6) upon an incoming call. Asterisk is returning 407 error. The Mediatrix does not support registration of its SIP usernames. How can I enable calls from Mediatrix to be accepted by Asterisk? Thank you in advance for your help, very much appreciated. Frame 46 (796 bytes on wire, 796 bytes captured) Ethernet II, Src: 192.168.0.27 (00:90:f8:00:ef:d1), Dst: 192.168.0.6 (00:0c:29:4e:99:37) Internet Protocol, Src: 192.168.0.27 (192.168.0.27), Dst: 192.168.0.6 (192.168.0.6) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.0.27;branch=z9hG4bKcac751873 Content-Length: 243 To: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=f0dfa5e35b9ce15 Call-ID: [EMAIL PROTECTED] CSeq: 1103931476 INVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Contact: Port 1 sip:[EMAIL PROTECTED] Supported: replaces User-Agent: MxSipApp/4.4.13.88 MxSF/v3.2.7.38 Message body Frame 47 (537 bytes on wire, 537 bytes captured) Ethernet II, Src: 192.168.0.6 (00:0c:29:4e:99:37), Dst: 192.168.0.27 (00:90:f8:00:ef:d1) Internet Protocol, Src: 192.168.0.6 (192.168.0.6), Dst: 192.168.0.27 (192.168.0.27) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Status-Code: 407 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.0.27;branch=z9hG4bKcac751873;received=192.168.0.27 From: sip:[EMAIL PROTECTED];tag=f0dfa5e35b9ce15 To: sip:[EMAIL PROTECTED];tag=as5d1a1ce8 Call-ID: [EMAIL PROTECTED] CSeq: 1103931476 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=7a237869 Content-Length: 0 Frame 48 (360 bytes on wire, 360 bytes captured) Ethernet II, Src: 192.168.0.27 (00:90:f8:00:ef:d1), Dst: 192.168.0.6 (00:0c:29:4e:99:37) Internet Protocol, Src: 192.168.0.27 (192.168.0.27), Dst: 192.168.0.6 (192.168.0.6) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[EMAIL PROTECTED] SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.0.27;branch=z9hG4bKcac751873 Content-Length: 0 To: sip:[EMAIL PROTECTED];tag=as5d1a1ce8 From: sip:[EMAIL PROTECTED];tag=f0dfa5e35b9ce15 Call-ID: [EMAIL PROTECTED] CSeq: 1103931476 ACK User-Agent: MxSipApp/4.4.13.88 MxSF/v3.2.7.38 Frank Attard Is Your Website Hackable? Check with Acunetix Web Vulnerability Scanner FREE trial version - http://www.acunetix.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NOTIFY Problem
Hi,one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function.Any idea? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Mediatrix 1204
Frank Attard wrote: I am pasting 3 SIP messages between the Mediatrix (192.168.0.27) and Asterisk (192.168.0.6) upon an incoming call. Asterisk is returning 407 error. 407 is not an error. SIP errors are in the 5xx and 6xx range. 407 means Asterisk is expecting the SIP device to provide authentication information. If you don't want that to be done, then you will need to tell Asterisk to use 'insecure' mode in sip.conf. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial 'R' option gone?
Benoit Panizzon wrote: Hi After migrating from 1.2.4 to 1.2.5 I noticed that: show application dial does not show the 'R' option anymore. Has this become an undocumented feature I'm running 1.2.7.1 and I do show the 'r' option. I would suggestion you remove the /usr/lib/asterisk/modules and re-install. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue Transfer
Dinesh the agents they receive a call and this call will have to be transferred, them uses only functions hold and trnsf in device Polycom IP 301, below mine features.conf This problem, only occurs with calls that if they originate in the pilot of queue and when an agent receives and transfers. It could help in this case me? Greetings Josué features.conf ;; Sample Parking configuration; [general]parkext = 700 ; What ext. to dial to parkparkpos = 701-720 ; What extensions to park calls oncontext = parkedcalls ; Which context parked calls are in parkingtime = 90 ; Number of seconds a call can be parked for ; (default is 45 seconds);transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8 ; Configure the pickup extension. Default is *8atxfer = #2 ; Transfer 2006/4/29, Dinesh Nair [EMAIL PROTECTED]: On 04/29/06 10:06 Josué Conti said the following: is that if the agent transfers the call, for another user and this user takes care of the call, the status of the agent in the show agents is of that it the same continues speaking (talking to zap) with circuithow are you performing the transfer ? are they blind/attended transfers using the keystrokes in features.conf ?--Regards, /\_/\ All dogs go to heaven.[EMAIL PROTECTED](0 0) http://www.alphaque.com/+==oOO--(_)--OOo==+| for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do || echo The opinions here in no way reflect the opinions of my $a $b.|| done; done| +=+___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTIFY Problem
Il Neofita wrote: Hi, one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '*MailScanner warning: numerical links are often malicious:* 192.168.100.124 http://192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function. Any idea? If you remove the mailbox= bit of sip.conf for that host, then asterisk will almost never send it a notify. Either way, is it causing a problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk to use an outbound proxy
Dear all, Do anyone know to setup asterisk's SIP channel to use an outbound proxy outside of asterisk's network to proxy the SIP message? Thanks Ray -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to use an outbound proxy
Raymond Chen wrote: Do anyone know to setup asterisk's SIP channel to use an outbound proxy outside of asterisk's network to proxy the SIP message? This is documented in the sample sip.conf file in the configs directory of your Asterisk source tree. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 28 Apr 2006, Klaus Darilion wrote: Back to ISDN BRI crossover cable. After reading some ISDN specs I came to the conclusion a crossover cable should be: 3---4 4---3 5---6 6---5 Yes. But I also found other pin layouts (e.g. http://www.cisco.com/warp/public/788/signalling/bri_voice_port_cfg.html) I also noticed these other layouts, but e.g. this does not make any sense to me. A BRI port layout is 3 2a 4 1a 5 1b 6 2b and the cisco layout would then cross '-' and '+' as well, not just the receive/transmit sides. Maybe this is necessary for cisco phones? I cannot explain that. Armin, how do you construct your BRI crossover cables? I use two possibilities. a) when I want to connect the ISDN card directly with the device using a short cable, I just cut the cable in the middle an reconnect them crossed and add the resistors here. (Maybe this is not the correct way for termination, but it always worked perfectly. b) when there is a ISDN bus (cables and plug wall mounted), I just add the cross and terminations to these boxes on the wall. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: NuFone Update: DIDs
Indeed... I just don't understand Nufone... we provide VoIP services... and have a contract inplace with our CLEC, we also have backup sources for numbers, LD termination, etc. No backup plan = BAD! On 4/28/06, Kerry Garrison [EMAIL PROTECTED] wrote: AMEN!! Any consultant that DOESNT take this into consideration should stick to installing Windows and calling themselves an IT Expert. You can screw up someone's network, mess up a workstation, hose their email, but you break someone's telephone service there will be hell to pay. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Friday, April 28, 2006 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: NuFone Update: DIDs Exactly why I chose to go with a PRI for business use. There is something to be said for the stability of a telco, even if it's not SBC (or ATT now). In some cases, government interference is good. How many businesses can survive a loss of their phone number? I know the ones I deal with cannot. This is something we need to take into consideration as Asterisk users or consultants. We need to look at the whole picture, not just a short term savings. Can your business/clients survive without listings in directory assistance or the phone books? Can they survive if they have to change numbers due to their Voip provider losing a contract? Some can, most can't. I looked into using Voip. Technically, it seemed like a good solution. I just don't trust it in the long run. I know that by using a telco, I will have access to my phone numbers. With Voip providers, who controls the numbers? I think this is something a lot of people fail to take into consideration. On 4/28/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I would be very wary, as VOIP providers feel no responsibility to the customer and will not bother to tell us they might not be around next week. Once bitten... -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual Timing Sources
Well that's what I did and they seem to be operating just fine. The CLEC told me even though they are the same CLEC, it is a different switch.. but yeah hehe in theory I guess the timing would have to be the same since THEIR switches are linked, eh? On 4/28/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: The following information is accurate for a situation where both T1s are connected to telco switches and the telco is therefore providing the timing signals. If one of the T1s is point to point (such as a tie line) then this information may or may not apply depending upon what's on the remote end. Configuring one of your T1s as the primary timing source tells Asterisk to sync to that T1's data stream as a reference clock for all data moving in and out of the system. Since both T1s are from the same provider, all clocks will be in sync. Even T1s from different providers are going to be pretty much in sync. You should be able to pick either T1 as your primary clock source. You can have only one primary clock source. The secondary timing source only comes into play when the first one fails. If the T1 that is your primary clock source fails, then Asterisk will begin using the secondary source. Without a secondary clock source configured, most equipment would instead switch to an internal clock source which would cause slips on the non-failed T1. Configuring a secondary clock source can help to ensure that you don't lose both T1s just because the primary fails. If possible, you should select whichever T1 is least likely to fail as your primary clock source. If there is no way to determine that, then just pick one. If you are interested in some detailed information about what timing is and how it works, this link looks pretty good: http://www.oreilly.com/catalog/t1survival/chapter/ch05.html Ok.. I sort of lied.. it's the same CLEC but two different switches.. was told by them since they are different switches I needed primary timing so in theory it should work if I set it as secondary.. ok we'll try! Just out of curiousity.. what happens if I set both as 1 (primary?) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial 'R' option gone?
Benoit Panizzon wrote: On Friday 28 April 2006 15:32, Eric ManxPower Wieling wrote: What does the R option do? Indicate 'Ringing' as soon as the called party indicates 'Ringing'. The 'r' option indicates 'Ringing' as soon as the connection is built, even if the called party is not yet ringing. With some SIP Services I then had the situation that the call was 'hanging' on the gateway, gut the caller heard a ringing, so tought the called phone would be ringing which was not the case. Apparently R also works with 1.2.5 even if not documented. R is not the same as r. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite, bandwidth, dial option
I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T, h, H, w, W or L (with multiple arguments). Probably there are more. I had in my memory that r, R, m would also prevent a reinvite. Can anybody say something on that? Below is a list of all options. o *t*: Allow the /called/ user to transfer the call by hitting # o *T*: Allow the /calling/ user to transfer the call by hitting # o *r*: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. o *R*: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff http://www.voip-info.org/wiki/index.php?page=Asterisk+zaphfc. o *m*: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. o *n*: (Asterisk 1.1 and later) July 2005 bug 752 http://bugs.digium.com/view.php?id=752 was included in CVS (Asterisk 1.1) and enhances the privacy manager considerably. As part of this patch, the 'n' flag to Dial got changed to be used as part of the privacy features, instead of being the 'dont jump to +101' flag. That flag is now 'j'. o *o*: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number) o *j*: Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels were busy (just like behaviour in Asterisk 1.0.x) o *M(*/x/*)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers) o *h*: Allow the callee to hang up by dialing *** o *H*: Allow the caller to hang up by dialing *** o *C*: Reset the CDR (Call Detail Record) for this call. This is like using the NoCDR http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+NoCDR command o *P(*/x/*)*: Use the PrivacyManager http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager, using /x/ as the database (/x/ is optional) o *g*: When the called party hangs up, exit to execute more commands in the current context. o *G(context^exten^pri)*: If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2). o *A(*/x/*)*: Play an announcement (/x/.gsm) to the called party. o *S(*/n/*)*: Hangup the call /n/ seconds AFTER called party picks up. o *d*: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also RetryDial http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+RetryDial o *D(*/digits/*)*: After the called party answers, send /digits/ as a DTMF stream, then connect the call to the originating channel. o *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) + *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play sounds to the caller. + *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the callee. + *LIMIT_TIMEOUT_FILE* - File to play when time is up. + *LIMIT_CONNECT_FILE* - File to play when call begins. + *LIMIT_WARNING_FILE* - File to play as warning if 'y' is defined. If *LIMIT_WARNING_FILE* is not defined, then the default
[Asterisk-Users] Telephone support charging system with Asterisk?
Hi, I'm interested in anybody that is providing a phone support service using an Asterisk system, with built in charging system. I run a PC support company and use Asterisk at the home/office. I would like to be able to provide technical support to my customers using asterisk. However I want to be able to charge them fairly for this support, and with little work on my part. My idea was for them to phone or login to a website and create a support account. They can then top this account up with X amount of credits, lets say 1 credit= 5 mins of support. Their account has a PIN associated with it. When they call to get support they have to enter there number and are told how much credits they have remaining. They then get put through to my office phone, if I am available and pickup, they start to get charged for the duration of their call. If I am unavailable they go to voicemail and dont get charged. Does this sound like something which is possible with Asterisk? Anybody doing something similar? Regards Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems if GXP-2000 phones and Asterisk are not on the same network
Mimmus wrote: Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? We have GPX-2000s connecting via different networks with no problems. However, at one point I had a real struggle to get them to register on certain lines but not others. The solution was to do a complete reset, wiping all settings and starting again. They now work OK most of the time. I'm using the current beta firmware (not alpha). Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How many asterisk process's are normal?
Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a production machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip): This is what I will eventually move over to since it is supported more. This machine is on a different network and not connected to the other test bed at all. What I found is that on test bed #2, I have 18 processes of asterisk running with absolutely no soft-phones connected to it, and 2 processes of mpg123 on hold music running. [EMAIL PROTECTED]:/usr/local/src# ps -aef |grep asterisk |wc -l 18 [EMAIL PROTECTED]:/usr/local/src# [EMAIL PROTECTED]:/usr/local/src# uptime 08:57:55 up 2 days, 6:12, 1 user, load average: 0.00, 0.00, 0.00 On Test bed #1, I have exactly 1 process of asterisk running with no phones connected to it. [EMAIL PROTECTED]:~ ps -aef |grep asterisk |wc -l 1 [EMAIL PROTECTED]:~ [EMAIL PROTECTED]:~ uptime 8:57am up 6 day(s), 8:19, 1 user, load average: 0.00, 0.00, 0.01 Nothing earth shattering, but strange. I assuming that 1 process at idle (no phones making calls and such) should be normal right? Terrelle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many asterisk process's are normal?
T.S wrote: Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a production machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip): This is what I will eventually move over to since it is supported more. This machine is on a different network and not connected to the other test bed at all. What I found is that on test bed #2, I have 18 processes of asterisk running with absolutely no soft-phones connected to it, and 2 processes of mpg123 on hold music running. [EMAIL PROTECTED]:/usr/local/src# ps -aef |grep asterisk |wc -l 18 [EMAIL PROTECTED]:/usr/local/src# [EMAIL PROTECTED]:/usr/local/src# uptime 08:57:55 up 2 days, 6:12, 1 user, load average: 0.00, 0.00, 0.00 On Test bed #1, I have exactly 1 process of asterisk running with no phones connected to it. [EMAIL PROTECTED]:~ ps -aef |grep asterisk |wc -l 1 [EMAIL PROTECTED]:~ [EMAIL PROTECTED]:~ uptime 8:57am up 6 day(s), 8:19, 1 user, load average: 0.00, 0.00, 0.01 Nothing earth shattering, but strange. I assuming that 1 process at idle (no phones making calls and such) should be normal right? Terrelle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This has been discussed a lot, in fact someone brought it up lastnight on the #asterisk IRC channel. Depending on a few factors your system will either show threads as other processes, or not at all. So essentially on some systems more then one Asterisk process will show up, and on others only one. As long as your Asterisk is running happily - then it's fine, multiple threads are started. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many asterisk process's are normal?
T.S wrote: Hello all, I have two test beds running the exact same version of asterisk 1.2.7.1, latest of zaptel, libpri, etc.. Test bed #1 (Solaris 9,sparc ultra 5): This one is closer to a production machine, in that it is connected to a sip provider thru an iax2 connection and have an incoming DID configured. I can send and receive calls. Test bed #2 (Slackware Linux 10.2, AMD XP chip): This is what I will eventually move over to since it is supported more. This machine is on a different network and not connected to the other test bed at all. What I found is that on test bed #2, I have 18 processes of asterisk running This gets hashed over every other month, you should check the archives. It's how your OS displays threads. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Locate Me Function with freePBX
The client's needs are the mother of invention. We have a client that currently uses a Cisco Call Manager and one of the features they love was the Locate-Me function (or follow-me, or find-me, whatever you want to call it) which basically rings their desk phone a few times then plays a short message, and then rings their other remote phones and cell phones. The customer wants this same functionalty from an Asterisk system that we will be building running freePBX. It took me a while to think about how to implement this without mucking around with the config files but the end result is a fairly simple solution that enables the use to turn on/off the function at will. For the complete article on how to implement this feature, go to:http://voipspeak.net/index.php?option=com_contenttask=viewid=72 Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTIFY Problem
I agree with you, but I would like to find a way to use the notification. I tough that there was a work around.On 4/29/06, tom [EMAIL PROTECTED] wrote:Il Neofita wrote: Hi, one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '*MailScanner warning: numerical links are often malicious:* 192.168.100.124 http://192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function. Any idea?If you remove the mailbox= bit of sip.conf for that host, then asterisk will almost never send it a notify.Either way, is it causing a problem?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telephone support charging system with Asterisk?
Mike Dent wrote: My idea was for them to phone or login to a website and create a support account. They can then top this account up with X amount of credits, lets say 1 credit= 5 mins of support. Their account has a PIN associated with it. This is a typical prepaid system at work. When they call to get support they have to enter there number and are told how much credits they have remaining. They then get put through to my office phone, if I am available and pickup, they start to get charged for the duration of their call. If I am unavailable they go to voicemail and dont get charged. A few lines in the dial plan or a macro is all that's needed for this. Does this sound like something which is possible with Asterisk? Anybody doing something similar? I don't do this for support since support is free for our customers. But you can use ASTCC and generate a prepaid card for each support account. Of course, you can call the number a support account number and they won't know the difference. :) You will of course have to take into account your e-commerce system to allow your customers to pay online. ASTPP works with OSCommerce while I've managed to make ASTCC work with Virtuemart/Joomla. There are other applications. Look around. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio Muting at seemingly random times
Hi everyone, I've been trying to chase down a bug in Asterisk 1.2. I have 2 completely different setups which exhibit the same problem, and I'm not sure what it is. For instance, one machine is setup as a voicemail server. If you call it, it says password..you put it in, and it says, You have On... ssage. Press 1 for new messages. Press 2 to ch..ge folders etc... One is a P III 700 with 2 X100P cards, and the other is a PII 450 pure SIP using either an X100P as timing, or ztdummy as timing, it exbits the same problem. With chan_zap not loading at all on the second box timing issues are all over the place. I also tested on a P4 2.8 Ghz Xeon and again, with no zap channels it's completely messed up. I was having trouble with the X100Ps on this machine, as it would cause the machine to throw an NMI, and reboot (This is an IBM X335). As a side note, I had NONE of these problems with 1.0.3 - 1.0.7. Two of the machines are running Debian, one is Fedora Core 4 (PII 450), all running 2.6 series kernels. Thanks for any ideas, -Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk dialing
Hi, I will try that thanks.Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] USB conference phone
I bought one of these. It's a great device. So good that I gave it to my boss to use with Skype. It's far better than the speakerphone in the Alcatel phone on his desk. We've used it with Skype and Gizmo. Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262 Original Message Subject: RE: [Asterisk-Users] USB conference phone From: Kerry Garrison [EMAIL PROTECTED] Date: Wed, April 26, 2006 7:01 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com it was a revewiers sample that I begged them to not make me send it back and they let me keep it. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, April 26, 2006 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] USB conference phone Lol now the important question.Did you pay for it or was it a reviewers sample J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, 26 April 2006 7:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] USB conference phone Yes, I have that device, I wrote the review of it and have used it regularly ever since. I use it with IDEFISK softphone for the most part but have tested it with Skype, X-Lite, and SJPhone. I have had it since November and just love it. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, April 26, 2006 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] USB conference phone Kerry, do you actually own one? Have you used it for long? What are you using it for? (jim personally I cant see the point of using your phone when I have a very good quality headset and mic.). Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, 26 April 2006 10:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] USB conference phone This is an excellent USB speakerphone http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Wednesday, April 26, 2006 6:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] USB conference phone I don't know about this phone but I can tell you I have a vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm It operates nice and has very good call quality. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, April 25, 2006 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to Make Asterisk-addons
The following occurs during make asterisk-addons.
I'm ok with asterisk but debugging things like this isnt my strong point.
Can anyone give me a pointer?
Thanks
Dan Journo
[EMAIL PROTECTED] src]# cd asterisk-addons[EMAIL PROTECTED] asterisk-addons]# makemake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara tions -D_REENTRANT -D_GNU_SOURCE -O6 -c -o format_mp3.o format_mp3.c
In file included from /usr/include/asterisk/logger.h:28, from /usr/include/asterisk/lock.h:83, from format_mp3.c:20:/usr/include/asterisk/compat.h:20: error: syntax error before __extension__
/usr/include/asterisk/compat.h:20: error: syntax error before '' tokenIn file included from /usr/include/asterisk/utils.h:36, from /usr/include/asterisk/cdr.h:48, from /usr/include/asterisk/channel.h:113,
from format_mp3.c:21:/usr/include/asterisk/strings.h:264: error: syntax error before __extension__/usr/include/asterisk/strings.h:264: error: syntax error before ';' token/usr/include/asterisk/strings.h:264: error: `__len' undeclared here (not in a fu nction)
/usr/include/asterisk/strings.h:264: error: initializer element is not constant/usr/include/asterisk/strings.h:264: error: syntax error before if/usr/include/asterisk/strings.h:264: error: redefinition of `__retval'
/usr/include/asterisk/strings.h:264: error: `__retval' previously defined here/usr/include/asterisk/strings.h:264: error: syntax error before const/usr/include/asterisk/strings.h:264: error: syntax error before '}' token
/usr/include/asterisk/strings.h:280: error: conflicting types for `strtoq'/usr/include/stdlib.h:346: error: previous declaration of `strtoq'format_mp3.c:46: error: redefinition of `struct ast_filestream'format_mp3.c:325: warning: function declaration isn't a prototype
format_mp3.c: In function `load_module':format_mp3.c:336: warning: passing arg 1 of `ast_format_register' from incompati ble pointer typeformat_mp3.c:336: error: too many arguments to function `ast_format_register'
format_mp3.c: At top level:format_mp3.c:342: warning: function declaration isn't a prototypeformat_mp3.c:347: warning: function declaration isn't a prototypeformat_mp3.c:359: warning: function declaration isn't a prototype
format_mp3.c:365: warning: function declaration isn't a prototype{standard input}: Assembler messages:{standard input}:58: Error: symbol `__retval' is already definedmake[1]: *** [format_mp3.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3'make: *** [format_mp3/format_mp3.so] Error 2
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[Asterisk-Users] Large Asterisk with Regexten, Regcontext, DUNDi, , , , , , , , , but not load balance...
Hi,all.
Have read a lots of documents and wiki and topic there.. I get a solution for Large asterisk...
1,in IAX or SIP config file...set..
[general]
regcontext = iaxregistrations
[peer]
name=peer
regexten= 10001
2,in extensions.conf.
[default]
exten = _X,1,Macro(dundi-priv,${EXTEN})
include = iax-clt
[iax-clt]
include = iaxregistrations
exten = _X,2,Answer
exten = _X,3,Dial(IAX/${EXTEN})
exten = _X,4,Hangup
[dundi-priv-local]
include = iax-clt
[dundi-priv-switch]
switch = dundi/dundi-priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten= s,1,goto(${ARG1},1)
include = dundi-priv-lookup
3,in dundi.conf
[mappings]
dundi-priv = dundi-priv-local,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER},nounsolicited,nocomunsolicit,nopartial
[peer]
...
include=dundi-priv
permit=dundi-priv
It can use realtime DB share IAX peer registration information and common extension information...
any advice and improvementwould be greate..-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150 -- Jefferyiaxtel Num: 1-700-576-1311
fwdnet Num: 728150
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[Asterisk-Users] RE: Install/Upgrade
Hi all, I was just wondering ifanyone knows of any gotchas with respect to upgrading Asterisk to the latest 1.2.7 ? Is the procedure the same? Config files remain intact? Just untar/make install? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
I have searched google and came up with too many options and packages that may or may not work for my needs, most articles seem to be for setting up routers. Maybe someone on the list can give me some better insight. I have monitoring turned on my shift eight (tm) (Asterisk ;-)) box for all calls. We have over one hundred agents and tons of recordings in wav format. I also have a cron job that runs a script to mux the in and out files and ftp them to a NAS device and it runs every five minutes. The NAS device and the * box are both directly connected to a Cisco Gigabit switch. I have had complaints of calls fading in and out and also cutting off. After reviewing the recordings, some of these complaints seem valid and I suspect the sheer bandwidth of the FTP traffic is causing the issues. I also run nagios checks on the box and get ping warnings on a regular basis. My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. I would like to accomplish throttling FTP on the Linux box with a solution that is not too elegant since this is a production machine in a busy call center. If I cannot do it on the * box I guess my next step is to see if the Cisco Gigabit switch has any QoS functionality. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Install/Upgrade
Make clean, make make install. Just dont do make samples. Dave Morrow wrote: Hi all, I was just wondering if anyone knows of any gotchas with respect to upgrading Asterisk to the latest 1.2.7 ? Is the procedure the same? Config files remain intact? Just untar/make install? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 / Lead, follow or get out of the way! / // This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] __mailto:[EMAIL PROTECTED]_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
It's a little crude but you can 1: Use VLAN(ing) on the Cisco Switch to segment the traffic on an addition 'LAN'. 2: Low Budget, Add a NIC on a separate network with the NAS. 3: Give me a bit, It'll come to me! :-) SNIP!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: NuFone Update: DIDs
I would not write a contract with a company that had these types of issues http://voxilla.com/name-News-article-sid-166.html Who eats $450,000? Matt wrote: Indeed... I just don't understand Nufone... we provide VoIP services... and have a contract inplace with our CLEC, we also have backup sources for numbers, LD termination, etc. No backup plan = BAD! On 4/28/06, Kerry Garrison [EMAIL PROTECTED] wrote: AMEN!! Any consultant that DOESNT take this into consideration should stick to installing Windows and calling themselves an IT Expert. You can screw up someone's network, mess up a workstation, hose their email, but you break someone's telephone service there will be hell to pay. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Friday, April 28, 2006 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: NuFone Update: DIDs Exactly why I chose to go with a PRI for business use. There is something to be said for the stability of a telco, even if it's not SBC (or ATT now). In some cases, government interference is good. How many businesses can survive a loss of their phone number? I know the ones I deal with cannot. This is something we need to take into consideration as Asterisk users or consultants. We need to look at the whole picture, not just a short term savings. Can your business/clients survive without listings in directory assistance or the phone books? Can they survive if they have to change numbers due to their Voip provider losing a contract? Some can, most can't. I looked into using Voip. Technically, it seemed like a good solution. I just don't trust it in the long run. I know that by using a telco, I will have access to my phone numbers. With Voip providers, who controls the numbers? I think this is something a lot of people fail to take into consideration. On 4/28/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I would be very wary, as VOIP providers feel no responsibility to the customer and will not bother to tell us they might not be around next week. Once bitten... -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
Alexander Lopez wrote: It's a little crude but you can 1: Use VLAN(ing) on the Cisco Switch to segment the traffic on an addition 'LAN'. The VLAN option would not work I dont think because the data is all going out the same interface whether or not it has a VLAN tag 2: Low Budget, Add a NIC on a separate network with the NAS. This is seriously being considered and may be the simplest and most effective way. This is an SGI Altix 350 so I am not very sure on what kind of cards it takes. I have never opened the case. 3: Give me a bit, It'll come to me! :-) Thanks for taking the time to help. SNIP!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Install/Upgrade
upgrading from what version? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Saturday, April 29, 2006 6:11 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] RE: Install/Upgrade Hi all, I was just wondering ifanyone knows of any gotchas with respect to upgrading Asterisk to the latest 1.2.7 ? Is the procedure the same? Config files remain intact? Just untar/make install? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
At 06:17 PM 4/29/2006, you wrote: My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. A really cheesy solution would be a second Ethernet card for the file transfers set to 10mb mode. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
Ira wrote: At 06:17 PM 4/29/2006, you wrote: My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. A really cheesy solution would be a second Ethernet card for the file transfers set to 10mb mode. Ira Not sure if cheesy is the right word. Sound solution may be a better adjective. Adding two NICs, one to each machine and connecting them directly via crossover cable on a totally separate network may be my best solution. No FTP traffic would even hit the NIC or the network used for VoIP and everything else. Unless there is a setting in Linux somewhere (still holding out hope) Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] = (Sound File Playback Application) == Registered application 'Playback' [app_dumpchan.so] = (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] = (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 7 == Registered translator 'lintoilbc' from format slin to ilbc, cost 245 [codec_g729a.so]Apr 29 22:25:25 WARNING[16253]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied Apr 29 22:25:25 WARNING[16253]: loader.c:554 load_modules: Loading module codec_g729a.so failed! Ouch ... error while writing audio data: : Broken pipe uname -a (Updated the hostname from the output.) Linux asteriskserver.XX.XXX 2.6.16-1.2096_FC5 #1 Wed Apr 19 05:14:36 EDT 2006 i686 i686 i386 GNU/Linux I re-downloaded the codec and attempted the i686 and i586 version wiht no luck. md5sum codec_g729a.so 92b64cc5be4a3e622c91357b116d99e3 codec_g729a.so Thanks -Jason -- Jason A. Kates ([EMAIL PROTECTED]) Fax:208-975-1514 Phone: 212-400-1670 x2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec G729 no longer works.
What version of Asterisk are you using? If it's trunk then you'll have to wait for the G729 codecs to be rebuilt with the new loader changes. On 4/29/06 11:49 PM, Jason A. Kates [EMAIL PROTECTED] wrote: I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] = (Sound File Playback Application) == Registered application 'Playback' [app_dumpchan.so] = (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] = (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 7 == Registered translator 'lintoilbc' from format slin to ilbc, cost 245 [codec_g729a.so]Apr 29 22:25:25 WARNING[16253]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied Apr 29 22:25:25 WARNING[16253]: loader.c:554 load_modules: Loading module codec_g729a.so failed! Ouch ... error while writing audio data: : Broken pipe uname -a (Updated the hostname from the output.) Linux asteriskserver.XX.XXX 2.6.16-1.2096_FC5 #1 Wed Apr 19 05:14:36 EDT 2006 i686 i686 i386 GNU/Linux I re-downloaded the codec and attempted the i686 and i586 version wiht no luck. md5sum codec_g729a.so 92b64cc5be4a3e622c91357b116d99e3 codec_g729a.so Thanks -Jason -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
On Sat, 2006-04-29 at 21:17 -0400, Steve Totaro wrote: My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. Is it the network/interrupt load or the CPU/RAM load that's causing the issue? If it's the later, seems like your SOL. If it's the former, I wonder if you could fiddle with traffic shaping in iptables to keep the FTP traffic down. Just a thought. Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- On site at GDOT's W.Annex, 404-463-2860 x199 -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT]Cisco 2621XM with (2) T1/PRI inetrfaces for sale
I know someone will suggest this should go on the -biz list, but this is a one time event and not a business for me. I have a new 2621XM router with (1) NM-HDV (1) VWIC-2MFT-T1 (4) PVDM-12 DSP modules (1) ADSL WIC (1) WIC-1DSU-T1 It was purchased for a prvate project that never got off the ground, and it taking up space. If anyone is interested, please contact me off list. Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec G729 no longer works.
This is the version reported on startup: Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. This is the list of packages I downloaded and compiled: asterisk-1.2.7.1.tar.gz asterisk-addons-1.2.2.tar.gz asterisk-sounds-1.2.1.tar.gz libpri-1.2.2.tar.gz zaptel-1.2.5.tar.gz Thanks -Jason On Sun, 2006-04-30 at 00:03 -0300, Joshua Colp wrote: What version of Asterisk are you using? If it's trunk then you'll have to wait for the G729 codecs to be rebuilt with the new loader changes. On 4/29/06 11:49 PM, Jason A. Kates [EMAIL PROTECTED] wrote: I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] = (Sound File Playback Application) == Registered application 'Playback' [app_dumpchan.so] = (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] = (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 7 == Registered translator 'lintoilbc' from format slin to ilbc, cost 245 [codec_g729a.so]Apr 29 22:25:25 WARNING[16253]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied Apr 29 22:25:25 WARNING[16253]: loader.c:554 load_modules: Loading module codec_g729a.so failed! Ouch ... error while writing audio data: : Broken pipe uname -a (Updated the hostname from the output.) Linux asteriskserver.XX.XXX 2.6.16-1.2096_FC5 #1 Wed Apr 19 05:14:36 EDT 2006 i686 i686 i386 GNU/Linux I re-downloaded the codec and attempted the i686 and i586 version wiht no luck. md5sum codec_g729a.so 92b64cc5be4a3e622c91357b116d99e3 codec_g729a.so Thanks -Jason -- Jason A. Kates ([EMAIL PROTECTED]) Fax:208-975-1514 Phone: 212-400-1670 x2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Large Asterisk with Regexten, Regcontext, DUNDi, , , , , , , , , but not load balance...
hi, all,,,
there have something need to correct
1,in IAX or SIP config file...set..
[general]
regcontext = iaxregistrations
[peer]
name=peer
regexten= 10001
2,in extensions.conf.
[default]
exten = _X,1,Macro(dundi-priv,${EXTEN})
exten = _X,2,Playback(invalid)
exten = _X,3,Hangup
;include = iax-clt ;delete this,, otherwise it would by pass do dundi-lookup//...
[iax-clt]
include = iaxregistrations
exten = _X,2,Answer
exten = _X,3,Dial(IAX/${EXTEN})
exten = _X,4,Hangup
[dundi-priv-local]
include = iax-clt
[dundi-priv-switch]
switch = dundi/dundi-priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten= s,1,goto(${ARG1},1)
include = dundi-priv-lookup
3,in dundi.conf
[mappings]
dundi-priv = dundi-priv-local,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER},nounsolicited,nocomunsolicit,nopartial
On 4/30/06, 陈帆 [EMAIL PROTECTED] wrote:
Hi,all.
Have read a lots of documents and wiki and topic there.. I get a solution for Large asterisk...
1,in IAX or SIP config file...set..
[general]
regcontext = iaxregistrations
[peer]
name=peer
regexten= 10001
2,in extensions.conf.
[default]
exten = _X,1,Macro(dundi-priv,${EXTEN})
include = iax-clt
[iax-clt]
include = iaxregistrations
exten = _X,2,Answer
exten = _X,3,Dial(IAX/${EXTEN})
exten = _X,4,Hangup
[dundi-priv-local]
include = iax-clt
[dundi-priv-switch]
switch = dundi/dundi-priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten= s,1,goto(${ARG1},1)
include = dundi-priv-lookup
3,in dundi.conf
[mappings]
dundi-priv = dundi-priv-local,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER},nounsolicited,nocomunsolicit,nopartial
[peer]
...
include=dundi-priv
permit=dundi-priv
It can use realtime DB share IAX peer registration information and common extension information...
any advice and improvementwould be greate..-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150 -- Jefferyiaxtel Num: 1-700-576-1311
fwdnet Num: 728150 -- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150
___
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[Asterisk-Users] Compare to Skype
One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine What indicates that there is no fault on his Internet connection!!! He is using his notebook and Xlite, but also tried the snom 360. Any hints? He also told me that he used another sip service before with the same bad result. I wonder if the Kaza boys have here something built in, bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NuFone - How to switch to another provider?
I have some DIDs from NuFone (tollfree). How can I switch them and to which provider? What is the cost for that? What is the procedure for that? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compare to Skype
One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine What indicates that there is no fault on his Internet connection!!! He is using his notebook and Xlite, but also tried the snom 360. Any hints? Is he calling you on another VoIP phone or calling you on a landline/cellphone (through the PSTN)? If he is calling a landline/cellphone, then it is probably your upstream termination provider that is having jitter problems (this is my exact problem). If I check my voicemails on my IP phone (which connects directly to my asterisk box 60 miles away), everything is great. HOWEVER, if I *dial* my telephone number and check my voicemails (as if I was calling in to check my voicemails), I get loads of jitter. So between my IP phone and my * box, the connection is great, but its what is after my * box that is causing the problem. Who is providing you termination? - Gabe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compare to Skype
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote: One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine What indicates that there is no fault on his Internet connection!!! He is using his notebook and Xlite, but also tried the snom 360. Skype uses iLBC codec, which has great jitter compensation. IIRC, the newer SIP channels of * are supposed to have the same capabilities, but I have not tested. I really do not like Skype (prefer FWD), but I must say, over satellite, etc, they provide quality.. All about the codec in this case.. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
Steve Totaro wrote: I have searched google and came up with too many options and packages that may or may not work for my needs, most articles seem to be for setting up routers. Maybe someone on the list can give me some better insight. I have monitoring turned on my shift eight (tm) (Asterisk ;-)) box for all calls. We have over one hundred agents and tons of recordings in wav format. I also have a cron job that runs a script to mux the in and out files and ftp them to a NAS device and it runs every five minutes. The NAS device and the * box are both directly connected to a Cisco Gigabit switch. I have had complaints of calls fading in and out and also cutting off. After reviewing the recordings, some of these complaints seem valid and I suspect the sheer bandwidth of the FTP traffic is causing the issues. I also run nagios checks on the box and get ping warnings on a regular basis. My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. I would like to accomplish throttling FTP on the Linux box with a solution that is not too elegant since this is a production machine in a busy call center. If I cannot do it on the * box I guess my next step is to see if the Cisco Gigabit switch has any QoS functionality. Thanks, Steve ___ Steve, If you don't want to get too fancy, you should switch to using rsync (if possible) and use the --bwlimit option. If you MUST use ftp, try using trickle: http://monkey.org/~marius/pages/?page=trickle I haven't used it, but you should be able to call your FTP upload binary (whatever it may be) with it and force a lower transfer speed. Let us know how it goes! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk
Steve Totaro wrote: I have searched google and came up with too many options and packages that may or may not work for my needs, most articles seem to be for setting up routers. Maybe someone on the list can give me some better insight. I have monitoring turned on my shift eight (tm) (Asterisk ;-)) box for all calls. We have over one hundred agents and tons of recordings in wav format. I also have a cron job that runs a script to mux the in and out files and ftp them to a NAS device and it runs every five minutes. The NAS device and the * box are both directly connected to a Cisco Gigabit switch. I have had complaints of calls fading in and out and also cutting off. After reviewing the recordings, some of these complaints seem valid and I suspect the sheer bandwidth of the FTP traffic is causing the issues. I also run nagios checks on the box and get ping warnings on a regular basis. My question is, how can I throttle the FTP (Standard with dist) transfers using out of the box CentOS4.3 (or any easy to use, low learning curve package)? I thought about FTPing the files at less frequent intervals but that just makes the issue less frequent but last longer. I would like to accomplish throttling FTP on the Linux box with a solution that is not too elegant since this is a production machine in a busy call center. If I cannot do it on the * box I guess my next step is to see if the Cisco Gigabit switch has any QoS functionality. Thanks, Steve Steve, Now for the fancy solutions: 1) Try enabling NAPI interrupt handling for your ethernet card. Some people report that it reduces interrupt load while increasing transfer speed by %85 - %100 (on the same processor, with the Intel e1000 driver and a good Intel NIC). I haven't tried it yet, but that is what I have read... 2) AstShape: http://www.krisk.org/astlinux/misc/astshape 3) AstShape and Cisco: http://www.krisk.org/astlinux/misc/astshape Configure the Catalyst to map packets with IP TOS 0x10 and 0x18 into the second highest priority (the highest priority is reserved for network control messages). Either whatever the Cisco has or 802.1p 5 (the highest priority for end user traffic). At gigabit speeds QoS configured like this will probably just waste CPU time (on both the machine and the switch) while not being very effective. Try my simple suggestions from my previous post first! :) -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Compare to Skype
Ronald Wiplinger wrote: One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. If that's what is going on (and other users don't have a problem), then it's likely to be a connectivity issue somewhere between the user's ISP and your own. A traceroute to his IP address may help you to identify the issue. He also told me that he used another sip service before with the same bad result. I wonder if the Kaza boys have here something built in, Perhaps the other sip service used the same ISP that you do... or his is somewhat flaky but happens to have a good connection to Skype... We keep our equipment in a very well-connected data center, because problems like that will kill your service and there's not much you can do to fix it (besides move). Yours, Yaakov Menken -- Yaakov Menken Capalon Internet Solutions Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problame with outbound calls on pri
Hi. recently I have been trying to setup a PRI on asterisk. Inbound calls are working just fine but I am not able to make outbound calls. Does anyone know what I need to change to make outbound calls work? Right now the PRI is instantly hanging up on the outbound calls. I have included full debug info as well as config files. /etc/zaptel.conf span=1,1,1,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us relevant portions of zapata.conf: [trunkgroups] [channels] language=en context=from-pstn pridialplan=unknown signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master ;signaling=em_w switchtype=4ess group=0 channel = 1-23 debug info: -- Executing Dial(SIP/202-2d92, zap/g0/18005551212) in new stack -- Making new call for cr 32771 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=48 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 02 80 90] Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 0f b1 74 65 73 74 20 65 78 74 65 6e 73 69 6f 6e] Display (len=15) Charset: 31 [ test extension ] [6c 05 21 83 32 30 32] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '202' ] [70 08 80 33 38 31 36 30 36 38] Called Number (len=10) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '18005551212' ] -- Called g0/18005551212 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE COMPLETE (90) [08 03 83 e0 20] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Transit network (3) Ext: 1 Cause: Unknown (96), class = Protocol Error (6) ] Cause data 1: 20 (32) -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup -- Channel 0/1, span 1 received AOC-E charging 0 units NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
