RE: [Asterisk-Users] oh323 configuration
Hi, To call the extensions registered on Asterisk. You don't need th gatekeeper. In your H.323 devices just set the gateway to Astiersk IP. I have test on ooh323 channel drive netmeeting. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Thursday, December 29, 2005 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] oh323 configuration El jue, 29-12-2005 a las 05:40 +0500, Rehan Ahmed escribió: Hi, What exactly would you like to do, how would you like asterisk to talk with GNUGK I'm a little confused about the use of ooh323. I want to register some elesign h.323 hardare with gnugk to call to sip devices conected with asterisk. It's possible ? Rehan On 12/28/05, Guillermo Salas M [EMAIL PROTECTED] wrote: It's possible to register oh323 with gnugk ? Any one knows one good oh323 how to? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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 -- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today. http://www.didx.net - DID Number Exchange and Peering Service. -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Getting Yoda unit to register all four ports
Hi Kevin, Now that makes perfect sense! Good call. I never got around to checking but the last time I worked with one of their tech help, he had an accel.com.tw email addy. No wonder the firmware commands were surprisingly familiar. As you probably know by now, they have never used *. The tech first tried to run * in their lab and failing that, later decided to remotely test around my unit. The units I bought were all originally H.323 before I switched to * I got the latest SIP firmware from Yoda and upgraded them myself. If you need help, let me know. I can re-enable my tftp server in Asia for you to use. The image file is it4mcs330.imz Consoleshow version Internet Telephony Gateway Version: gs020200140ena_0406mc Boot Loader Version: 4.13 RTOS Version:2.5.0/BE SIP Stack Version: 3.0.4.1 DSP image Version: 8.1.2.1. TSG Version: R8.0 Gateway (Build 4) Console Did you follow these same steps? Welcome to minicom 2.1 OPTIONS: History Buffer, F-key Macros, Search History Buffer, I18n Compiled on Oct 27 2004, 16:57:58. Press CTRL-A Z for help on special keys Incorrect password Password? Incorrect password Password? *** Consoleconfig erase The system configuration data will be erased from non-volatile storage permanently. Are you sure to erase it (y/n)? [n] y System configuration records erased from flash Consoledownload == WARNING == * Entering download mode will hang up all telephone connections * * and all the configuration settings will lose. * * Be certain all the configuration settings have been saved. * === Do you want to enter download mode now (y/n)? [n] y Boot loader V4.13 Mem 16b 16M Loading s/w upgrade utility. ** Internet Telephony Gateway TFTP Loader Ver 5.00 ** EITGLoaderstart Allocated 0x730200 bytes = 7360 KB for downloading files IP address of the TFTP server? [24.199.11.42] File name? it4mcs330.imz Starting download file: it4mcs330.imz 728K bytes read Download complete, file size = 751636 Application code downloaded successfully Do you want to write downloaded image to flash EEPROM (y/n)? [y] Press Enter to start flash EEPROM programming Flash EPROM programming on-going, BE CERTAIN NOT TO TURN POWER OFF... Programming Application code Sector 12 of 12 programmed Flash programming completed All sectors programmed successfully Download another file (y/n)? [n] EITGLoaderquit Do you want to restart the system now (y/n)? [n] y kevin ling wrote: Hi, I download the guide from yoda site. It's seems the original vendor is Accel AmiGate Elite 400 (http://www.accel.com.tw/frame/frame_age400.htm) I have one H.323 model and can't upgrade to SIP firmware. So what is your firmware version? Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Friday, December 30, 2005 5:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting Yoda unit to register all four ports I have a sample of the Yoda VG400 and I am having a devil of a time trying to get all four channels to register to Asterisk. I have an Asterisk 1.2.1 server. I have tried adding one at a time and rebooting it, but it stops after the first. http://www.yoda.com.tw/model.php?type=Enterprise_VoIPpname=VG400 Anyone had success with this? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- 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
Re: [Asterisk-Users] SoCal Users Group Meeting Schedule
SFVLUG also has a couple of * aware members. Sent a copy. Mike Fedyk wrote: Forwarded to OCLUG, LUGIE UUASC which have members that have expressed interest in asterisk. Mike Kerry Garrison wrote: The SoCal Asterisk Users Group will be meeting at the Heritage Park Public Library on the corner of Walnut and Yale in Irvine on the 3rd Thursday every month. The following dates are already secured: Thurs Jan 19 Thurs Feb 16 Thurs Mar 17 Irvine Heritage Park Library (949) 936-4040 14361 Yale Ave Irvine, CA 92604 Google Directions: http://tinyurl.com/9vq3e -- 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
Re: [Asterisk-Users] CDR problem - incorrect time
Chris Mason (Lists) ha scritto: We have a billing system that depends on the CDRs. We had a guest that made a one minute call to a local cellphone, this call went out Zap channel through our channel bank. The CDR recorded a 200 minute call, but I checked with the Telco's records and it had terminated after one minute. What can cause this and what can I do to prevent it? happened to me once, I've noticed that the txt cdr (under /var/log/asterisk) was missing the line for that call, so probably something went wrong writing that record, it's not a solution but at least double checking the db cdr with the txt ones is a way to look for such errors. ___ --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 application newbie help
Dear Paul H., Thanks my dear friend, that worked. Thanks a lot for the help. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] Nested MySQL Commands
Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Just make sure that you use different variable names for each query if the values should stay available after the next query. What we tend to do is grab the data from the database and the stuff that should stay around for a longer time is assigned to a new and appropriately named variable. So the original variable can be used again. We've got loads of queries in our extensions.conf. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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: Issue calling other PBX systems using VoIPwithPolycom 501
-Original Message- [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Thursday, January 12, 2006 4:37 AM This is supposedly fixed in 1.2.1. The issue is that some companies using PRI are starting their IVR as the ringback tone. The problem is that asterisk considers the call not answered yet, so will not send any DTMF tones. Some credit card companies, airlines and ATT conferencing numbers do this. They do this to save money. Their toll free vendor starts charging them for your call after they answer, so they save a few seconds on every call by not answering right away. I tried to hack chan_zap.c in my 1.0.9 installation to fix this, but was unable to. I'm having some similar problem with a toll free number with IVR in Italy. It rings indefinitely and Asterisk doesn't detect answer. A big issue is that I'm using Asterisk 1.2.1! Any help will be greatly appreciated. Mimmus ___ --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: RE : Re: RE : codecs order and so on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Calling zap = no problem, Ulaw is choosen Calling pstn provider =fail (I need g729 but Ulaw is choosen) Call from zap = no problem Ulaw is choosen Call from pstn = no problem g729 used... When you call out * establishes two channels. One is between Ua and *, and another between * and Zap (or provider). If you call out, asterisk first negotiate codec for that channel. Then it tries to nagotiate codec for second channel. When you call your provider it can't nagotiate because he doesn't have g729 codec. This is reason why you have problem, and I have explain how to solw it. There is nothing else I can say to help you. -- Tomislav Parcina [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] Re: RE : Re: RE : codecs order and so on
Thanks for all, But Asterisk is able to use g729 pass-tru and both ends have g729, then the question is: Why asterisk doesn't use g729 pass-thru when both ends have it? For incoming calls from Voip, G729 is not a problem, problems appears when I make a call to Voip... Olivier Ps: No need to answer, that's just a fact -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tomislav Parcina Envoyé : jeudi 12 janvier 2006 10:31 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Re: RE : Re: RE : codecs order and so on In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Calling zap = no problem, Ulaw is choosen Calling pstn provider =fail (I need g729 but Ulaw is choosen) Call from zap = no problem Ulaw is choosen Call from pstn = no problem g729 used... When you call out * establishes two channels. One is between Ua and *, and another between * and Zap (or provider). If you call out, asterisk first negotiate codec for that channel. Then it tries to nagotiate codec for second channel. When you call your provider it can't nagotiate because he doesn't have g729 codec. This is reason why you have problem, and I have explain how to solw it. There is nothing else I can say to help you. -- Tomislav Parcina [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] Nested MySQL Commands
Douglas Garstang ha scritto: Peter, I assume you mean something like this in extensions.conf: exten = _X.,1,AGI(master-dial-logic.pl) and then there's only one call. All logic would be performed by the perl script. This has many advantages. One disadvantage however is that potentially, there could be 120 simultaneous instances of this script running (one per call). Douglas. but you can use fastagi, it will be maybe a little more complex to write the server code but it should scale better, shouldn't it ? ___ --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] Nested MySQL Commands
Douglas Garstang ha scritto: So I really wish there was some way to measure how well the worst case scenario would perform. This would be 120 simultaneous calls (don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850 with 2GB RAM. Asterisk would call an AGI script, written in perl, to route all calls. The script would have to perform multiple database queries in order to route a call. It will work if you need no transcoding, I tested a python agi doing something like 6 query to accept / instradate the call and it works for 150 / 200 simultaneous calls, the machine starts sweating of course, but the voice quality is still good, no drops. Mine is just a quick prototype, using fastagi or writing the agi in C is surely the way to go, imho fastagi will let you have a more configurable / customizable system since you can write the application in a object oriented language. ___ --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] read .what else to do ?
Hi all , I have tried configuring Asterisk at home to make calls outside our Lan WITHOUT any success (Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT )To be precise i did the following (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 1 to 2 to 192.168.1.2 (2) I set externip = x.x.x.x (to our public WAN) localnet =192.168.1.0 /255.255.255.0 (3) I also set nat=yes qualify=yes (4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ).Please advise me how to make it work !!! (5) I am using xten lite soft phone on my pc .(6) I use cisco 1700 series router , and i have natting configured on this router .Maybe I am using a wrong command .Please,tell me the commands to forward the ports Port 5060-5082,1 to 2 to 192.168.1.2 on a cisco router . Please reply and advice !!! Thanks Yahoo! Photos Showcase holiday pictures in hardcover Photo Books. You design it and well bind it!___ --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] Avoided initial deadlock
Hi All, I am getting some error from the sip channel, anyone can tell me what is the meaning of this error, Is it some harmfull warnnings? I will be appricate if any one can help me. -- Thank You, Code Lover ___ --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] Catv ATA problem
The Cable TV company has given me a two line ATA/Modem device made by Arris. I figured on taking the FXS ports on it and connecting them to my channel bank as additional lines. When I tested the lines with an analog phone the line worked great, very clear and loud. When I connected to the channel bank, the calls were great until they terminated after about 30 seconds. A few more tries confirmed it was random, sometimes it hung up right away, sometimes after a few seconds. Any ideas why? The Channel bank was set to FXO_LS in it's configuration, and they confirmed the port is loopstart. Any help appreciated. -- Chris Mason ___ --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] Fwd: voip - forwarding ports
Note: forwarded message attached. Yahoo! Photos Showcase holiday pictures in hardcover Photo Books. You design it and well bind it!---BeginMessage--- I read with interest your question"Aastra 9133i and NAT: Can it work?" in Asterrisk user"I've got an Aastra 9133i (with the latest firmware version) and a Cisco7960 sitting behind a NAT device on my LAN. The Asterisk server is hosted offsite and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnelthe SIP messages initiated by the Asterisk box. It works like a charm with the Cisco phone by using the following config info: "I am in interested in how you configure your router to forward the port number .In a nut shell how can i configure my router CISCO ROUTER toFo rwarded UDP Port 5060-5082 to 192.168.1.2Forward UDP Port 1 to 2 to 192.168.1.2 . I will appreciate your response thank you . Thanks Yahoo! Photos Ring in the New Year with Photo Calendars. Add photos, events, holidays, whatever.---End Message--- ___ --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] Web based SIP client
Hi I found this http://www.etntalk.com/callto/loginany/ Somebody has used it? roberto2006/1/11, Derek Whitten [EMAIL PROTECTED]: Miguel wrote: Roberto Pereyra wrote: Hi Someone knows a free web based SIP client for use with any provider ? Thanks roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Roberto, im looking for a similar solution,i found this on the archives http://www.microappliances.com/site/html/index.php It seems very complete to me (look at the customers page), does anyone here have it in production? Any comment? thanks in advance --- Miguel ___ --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 There was someone here on the lists a while ago that had a java basediax client..might find it if you search the archives..--.-BEGIN GEEK CODE BLOCK- Version: 3.1GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK--. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Ing. Roberto Pereyra ContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 ___ --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] GSM codec problem - Windows messenger 5.1
Hi, Im using Windows Messenger 5.1 (It supports SIP) with Windows XP to connect to another SIP user using asterisk in the middle. The codec selected is gsm and I have a problem because the sound sent by my machine reach the end point all wrong, it seems just noise If I use uLaw (or gsm from other machine with othe UA) everything works fine. I think the audio codec used at my machine is the Microsoft GSM 6.10 As anyone experienced some problem like this one, or similar with this codec? Thanks, Ricardo ___ --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: chan_bluetooth problems
I power cycled my phone and this problem went away. However, it still doesn't work as expected. The call is initiated but the audio can't be heard through asterisk. If I pick up the cell phone I can carry on the conversation there. My phone is a Nokia 6255i Again any ideas? Thanks in advance --- Originally John Voss said Has anyone had experience with Theo's chan_bluetooth with asterisk-1.2.0 on FC3. When I try to make a call to my home through the channel my phone shows calling but my home phone never rings. I get this on screen: == Spawn extension (outgoing, 3234600, 2) exited non-zero on 'SIP/2299-9131' -- Executing Dial(SIP/2299-7743, BLT/John/3541107) in new stack [AG] John ATD3541107; -- Called John [AG] John OK [AG] John +CIEV: 3,2 [AG] John +CIEV: 4,2 [AG] John ERROR [AG] John +CIEV: 1,0 [AG] John ATH [AG] John AT+CHUP == No one is available to answer at this time (1:0/0/0) [AG] John +CIEV: 3,0 [AG] John +CIEV: 4,0 [AG] John OK [AG] John OK Any suggestions? Anyone know where there are any documentation for it. -- ___ Play 100s of games for FREE! http://games.mail.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] No D-channels available! Using Primary on channel 16 anyway!
Hi!I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and * 1.2.Every few hours I get this message and asterisk dies just after that:Warning: No D-channels available! Using Primary on channel 16 anyway!When this happens restarting zaptel and asterisk services, generally puts the system back onlinemy zaptel.con reads:span=1,1,0,ccs,hdb3 #span=2,0,0,cas,hdb3 span=3,2,0,esf,b8zs #-- This because we have two American CBsspan=4,3,0,esf,b8zsbchan=1-15dchan=16bchan=17-31fxoks=63-86fxoks=87-110loadzone = usIdeas anybody? Please? Things done:* zttool/ztcfg* Trying R2 instead of PRI (R2 is the south americanstatdar, which wont even start)*Added crc4 to span1, with ugly sound consequences-- Paavum Regina, Per Secula et Secularum!! ___ --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: FXS or VOIP
Hi Jim, My decision had more to do with the infrastructure of the existing wiring more than anything else. I really *wanted* to go with voip but I couldn't justify the extra cost since our office is wired for analog. I ended up going with the TE410P Quad span T1 card, 2 PRIs and an adit-600 channel bank for the FXS ports. I really had to do very little to tune the FXS ports other than setting tx and rx gain on the channel bank. We have 5 other branch offices that we are connected to via WAN and we have * servers at each of those locations, doing voip between those and also the larger install that I describe above. So just because you have FXS ports does not mean that you cannot do voip. There's always services like nufone for long distance that you can connect * to. For your smaller setup just evaluate what's there already in terms of network infrastructure then decide what fits best for both your budget and your growth. Best Regards, Jason Stewart On 11/01/06 15:06 -0600, Jim Freeze wrote: Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. Thanks -- Jim Freeze ___ --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] (no subject)
asterisk-users@lists.digium.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] Build Error - ZT_EVENT_DTMFDIGIT
Hi, I've seen a few posts about this but no fix. Anyone able to help? Here's what I did: I configured a brand new machine with Redhat 9.0. I made sure that I had: bison cvs gcc kernel-source libtermcap-devel ncurses-devel newt-devel openssl1096b openssl-devel readline41 readline-devel zlib zlib-devel When I went to get Asterisk I did the following: cvs checkout zaptel libpri and cvs checkout -r v1-0_stable asterisk I built zaptel then libpri and had no problem, but asterisk complains with the following error: chan_zap.c:2772: `ZT_EVENT_DTMFDIGIT' undeclared (first use in this function) How do I fix this? Thanks, Hugh ___ --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: Re: Recommend Fax Hardware for T1 PRI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Wonderful advice. Both of these solutions actually fail for most people. Digium card worked for me. -- Tomislav Parcina [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] Zaptel SVN
On 1/12/06, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote: Hi, i can't compile the latest svn update from zaptel: /lib/modules/2.6.14-1.1653_FC4smp/build make -C /lib/modules/2.6.14-1.1653_FC4smp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:6193:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:224: Warnung: »fcstab« definiert, aber nicht verwendet /usr/src/zaptel/zaptel.c:6193:5: warning: CONFIG_ZAPATA_DEBUG is not defined CC [M] /usr/src/zaptel/tor2.o CC [M] /usr/src/zaptel/torisa.o /usr/src/zaptel/torisa.c:1145: Warnung: »set_tor_base« definiert, aber nicht verwendet CC [M] /usr/src/zaptel/wcusb.o CC [M] /usr/src/zaptel/wcfxo.o CC [M] /usr/src/zaptel/wctdm.o CC [M] /usr/src/zaptel/wctdm24xxp.o CC [M] /usr/src/zaptel/ztdynamic.o CC [M] /usr/src/zaptel/ztd-eth.o /usr/src/zaptel/ztd-eth.c:185: Warnung: Initialisierung von inkompatiblem Zeigertyp CC [M] /usr/src/zaptel/wct1xxp.o CC [M] /usr/src/zaptel/wct4xxp.o /usr/src/zaptel/wct4xxp.c: In Funktion »t4_interrupt«: /usr/src/zaptel/wct4xxp.c:2219: nicht implementiert: »inline« beim Aufruf von »__t4_framer_interrupt« gescheitert: function body not available /usr/src/zaptel/wct4xxp.c:2251: nicht implementiert: von hier aufgerufen make[2]: *** [/usr/src/zaptel/wct4xxp.o] Fehler 1 make[1]: *** [_module_/usr/src/zaptel] Fehler 2 make[1]: Leaving directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686' make: *** [linux26] Fehler 2 It has just been fixed in /trunk -- Bird's The Word Technologies, Inc. http://www.btwtech.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] dCAp
HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk´s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and outbound routing, VoIP providers like voipjet, etc. Do you think the bootcamp is a good option???thanks ___ --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] Spandsp
I have tried to install spandsp. On fresh installed FC4 and Asterisk
1.2.1 with zaptel, addons and sounds. I have libtiff, libtiff-devel,
libxml2 and libxml2-devel RPMs installed.
I have untar spandsp-0.0.2pre22.tar.tar and have run
./configure
make
make install
then I have execute patch (at the end of mail) and I didn't recive any
error.
I have again run in /usr/src/asterisk-1.2.1/ dir
make clean; make; make install
and when I tried to start *, it fails when tries to load app_txfax.so.
What could be wrong?
[format_ilbc.so] = (Raw iLBC data)
== Registered file format iLBC, extension(s) ilbc
[app_curl.so] = (Load external URL)
== Registered custom function CURL
== Registered application 'Curl'
[EMAIL PROTECTED] /]#
*** patch file ***
--- Makefile.orig 2006-01-11 18:39:21.0 +0800
+++ Makefile2006-01-11 18:40:46.0 +0800
@@ -52,10 +52,14 @@
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h
$(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h
$(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),)
+APPS+=app_rxfax.so app_txfax.so
+endif
+
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L
$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
@@ -100,10 +104,16 @@
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} $(CURLLIBS)
+app_rxfax.so : app_rxfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
+app_txfax.so : app_txfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} -
L/usr/local/pgsql/lib -lpq
--
Tomislav Parcina
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Build Error - ZT_EVENT_DTMFDIGIT
hugolivude wrote: When I went to get Asterisk I did the following: cvs checkout zaptel libpri and cvs checkout -r v1-0_stable asterisk This is wrong. You do _not_ want to try to use CVS HEAD Zaptel and libpri with CVS v1-0 asterisk. Also, if you pull the v1-0_stable tag from CVS, you'll be getting a very old version. Why don't you start over with some proper checkout instructions as documented on the wiki... and maybe try to use Subversion (because it's easier) and version 1.2 of everything (because version 1.0 is no longer being updated)... you'll end up with a better system :-) ___ --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] conditional canreinvite
Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho, correct and logical), but, is possible to combine both, so do reinvite only within e.g. one context and disable reinvite when connecting phones between two context, or any better option exist/planned how to solve? thanks PJ ___ --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] Zaptel SVN
BJ Weschke wrote: It has just been fixed in /trunk That fix should be merged over to branch-1.2 as well then, if needed. ___ --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] OOH323 Configuration with Cisco FSX ports, no Gatekeeper
This customer has a simple 17xx unit with 2 2FSX VWIC modules. Wants to originate/terminate calls to my asterisk box as a last-ditch backup. There is no gatekeeper, needs to work solely with dial-peer statements on the Cisco side. I'm not an H323 guru, but I'm sure the problem is on the Cisco side as I get a cannot gateway iftype 1 to 1 for cid(28) debug message and no traffic ever reaches the asterisk box. Tim _ From: kevin ling [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 01:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OOH323 Configuration with Cisco FSX ports,no Gatekeeper Hi, You mean Cisco FXS Port? Can you describe more detail about your network configuration? Regards, Kevin _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy R. McKee Sent: Thursday, January 12, 2006 6:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OOH323 Configuration with Cisco FSX ports,no Gatekeeper Has anyone used the OOH323 driver to connect with the FSX ports on a Cisco router *without* the use of a Gatekeeper? If so could you share your OOH323 and Cisco configs? Thanks, Tim McKee File: ATT00246.txt attachment: winmail.dat___ --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] dCAp
On 1/12/06, blackgecko [EMAIL PROTECTED] wrote: HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk´s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and outbound routing, VoIP providers like voipjet, etc. Do you think the bootcamp is a good option??? As someone that took the dCAP exam, unless you're extremely comfortable with ALL aspects of Asterisk, I'd go for the Bootcamp. The dCAP written and practical covers pretty much everything about Asterisk. I was already doing some development with Asterisk and still found it challenging when I took it, so I'd recommend the bootcamp if you don't want to keep paying to retake the test. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Re: Re: Phones
It has worked very well. The MWI works when there is a VM. Their web interface lets you set any softkey to one of: Unassigned Line 1 Line 2 Line 3 Line 4 Line 5 Line 6 Line 7 Line 8 Line 9 Line 10 Line 11 Line 12 Line 13 Line 14 Line 15 Line 16 Line 17 Line 18 Line 19 Line 20 Line 21 Line 22 Line 23 Line 24 Line 25 Auto Answer Conference Do Not Disturb Handsfree/Mute Hold Message Waiting Mic Redial Release Transfer Park/Retrieve Pickup Speed-Dial I bought ours from them directly. I was able to negotiate the price with them. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Paul Dugas [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Wed, 2006-01-11 at 23:02 -0500, Steven wrote: Check out Citel.com . We are using their boxes to reuse NEC digital phones. I think that they had Lucent boxes as well. These connect via SIP to asterisk and have 24 digital ports. We used these because we already had the phones left over from a building we closed. While I have no use for this device at this time I think it could be *very* handy for some customers. Can you report on how well it works? How does it handle soft keys on the handsets? Any stability issues? I found one place online offering one for $3k ($125/port). Have you see better? Thanks, 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. ___ --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] Re: Nested MySQL Commands
Do you have a link to where it says this? The DBI docs that I looked at (perldoc dbi) said that it isn't thread-safe. -Original Message- From: Leo Ann Boon [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 12:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Douglas Garstang wrote: I also don't believe perl DBI is thread safe The lastest docs says that DBI does support multithread connection pooling. Otherwise, you are always free to implement your AGI in 'modern' :) programming languages like Java or C# that support threads and pooling. -Original Message- From: Douglas Garstang Sent: Wed 1/11/2006 9:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Re: Nested MySQL Commands Since about 1992... and the Asterisk docs for FastAGI are pretty rotten. But that's ok, I've come to expect that. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Wed 1/11/2006 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Douglas Garstang wrote: I don't get the whole concept of FastAGI. It's nothing special. Asterisk just opens a connection to a TCP port instead of executing a binary. How long have you been around Unix/Linux systems? Do you have any clue how much less expensive it is to open a TCP socket as compared to forking the Asterisk process, exec()-ing another program, having that program open database/web connections, etc.? ___ --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 ___ --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] Nested MySQL Commands
Andreas. I tried that. Still didn't work. It just appears that Asterisk doesn't like letting you execute another query while it's holding on to the state of a previous one. Doug. -Original Message- From: Andreas Sikkema [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 2:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Nested MySQL Commands Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Just make sure that you use different variable names for each query if the values should stay available after the next query. What we tend to do is grab the data from the database and the stuff that should stay around for a longer time is assigned to a new and appropriately named variable. So the original variable can be used again. We've got loads of queries in our extensions.conf. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] dCAp
On Thu, 2006-01-12 at 09:54 -0500, BJ Weschke wrote: As someone that took the dCAP exam, unless you're extremely comfortable with ALL aspects of Asterisk, I'd go for the Bootcamp. The dCAP written and practical covers pretty much everything about Asterisk. I was already doing some development with Asterisk and still found it challenging when I took it, so I'd recommend the bootcamp if you don't want to keep paying to retake the test. At one time someone posted that they were from AU and took the dCAP test in Europe and there were no questions about an E1 instead it was about T1s. What was your experience and location of the test? Are they regionalized? Further that person indicated they hadnt received (at the time of posting - unknown if this was ever resolved) a certificate of successful completion of the exam, nor listing on a webpage (they said it was promised that they would be listed). Did your experience differ? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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
Re: [Asterisk-Users] Build Error - ZT_EVENT_DTMFDIGIT
Thanks Kevin; exactly what I was looking for. I was using: cvs checkout -r v1-0_stable asterisk because that's what it said to do on the cardboard card that was packed with my TDM400. I wondered about the mixing and matching of HEAD and STABLE when I was writing the email, but that's all I could see to do when I was following the instructions given. I checked out the wiki. Just to confirm, if I issue the command: # svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I'd get the most recent _working_ version of Asterisk. But if I issue the command: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 I'd get the most recent _stable_ version of Asterisk. I'd be most grateful if you could confirm this form me. Many Thanks, Hugh On 1/12/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: hugolivude wrote: When I went to get Asterisk I did the following: cvs checkout zaptel libpri and cvs checkout -r v1-0_stable asterisk This is wrong. You do _not_ want to try to use CVS HEAD Zaptel and libpri with CVS v1-0 asterisk. Also, if you pull the v1-0_stable tag from CVS, you'll be getting a very old version. Why don't you start over with some proper checkout instructions as documented on the wiki... and maybe try to use Subversion (because it's easier) and version 1.2 of everything (because version 1.0 is no longer being updated)... you'll end up with a better system :-) ___ --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] FXS or VOIP
A factor no one here mentions is user psychological comfort. That's a great point. On my home setup the wife avoids using the SNOM's because it looks uber-intimidating and things like call transfer, park etc blows her mind, she doesn't get it. So I dusted off some Vista 350's I had and put them out in the house and she's much more comfortable with them. (although to her credit she is starting to use the SNOM's, just none of the fancy features. Funny, my kids think the SNOM's rock. They intercom each other and play with my custom *XX features all the time) This underscores the need in system design to design it for USERS. I should tatoo it on the back of my hand. ___ --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] Spandsp
makefile.patch is buggy. Compile app_rxfax and app_txfax by hand. Mimmus ___ --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] dCAp
I passed the dCap exam at Astricon last year without doing any of the training and it's not easy, it would be very difficult to pass without having practical asterisk knowledge. You really need to know your stuff. However if you have experience with all the things you listed you should be ok. I would suggest you do some background reading on voip history - eg h.323 and mgcp, standards and the asterisk cli. Make sure you know how to configure things like iax.conf, sip.conf, zaptel,conf, zapata.conf, meetme.conf etc etc Craig - Original Message - From: blackgecko [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 12, 2006 10:26 PM Subject: [Asterisk-Users] dCAp HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk´s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and outbound routing, VoIP providers like voipjet, etc. Do you think the bootcamp is a good option??? thanks ___ --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] Spandsp
Do you have the spandsp libraries in your library path?, by default they go
into /usr/local/lib
Craig
- Original Message -
From: Tomislav Parcina [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 12, 2006 10:32 PM
Subject: [Asterisk-Users] Spandsp
I have tried to install spandsp. On fresh installed FC4 and Asterisk
1.2.1 with zaptel, addons and sounds. I have libtiff, libtiff-devel,
libxml2 and libxml2-devel RPMs installed.
I have untar spandsp-0.0.2pre22.tar.tar and have run
./configure
make
make install
then I have execute patch (at the end of mail) and I didn't recive any
error.
I have again run in /usr/src/asterisk-1.2.1/ dir
make clean; make; make install
and when I tried to start *, it fails when tries to load app_txfax.so.
What could be wrong?
[format_ilbc.so] = (Raw iLBC data)
== Registered file format iLBC, extension(s) ilbc
[app_curl.so] = (Load external URL)
== Registered custom function CURL
== Registered application 'Curl'
[EMAIL PROTECTED] /]#
*** patch file ***
--- Makefile.orig 2006-01-11 18:39:21.0 +0800
+++ Makefile 2006-01-11 18:40:46.0 +0800
@@ -52,10 +52,14 @@
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h
$(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h
$(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),)
+APPS+=app_rxfax.so app_txfax.so
+endif
+
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L
$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
@@ -100,10 +104,16 @@
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} $(CURLLIBS)
+app_rxfax.so : app_rxfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
+app_txfax.so : app_txfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} -
L/usr/local/pgsql/lib -lpq
--
Tomislav Parcina
[EMAIL PROTECTED]
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RE: [Asterisk-Users] MTU and Voice Delay (latency??)
Rich Adamson wrote: So if I leave it as is (both set to Auto) then Flow Control is Disabled on the 3COM switch If I configure it so the Flow Control is Enabled then the 3COM defaults to Half Duplex. Is there a way for you to use ethereal to see what's coming through the dsl circuit? What happens if you set the 3com AND Speedtouch to full duplex? Flow control is not necessary for what you're doing; disable it if you can. I will attempt to set both to 100Full after hours UK time today. As for ethereal, I have several captures, what in particular would you recommend I look for? Thanks, Geoff ___ --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: RE : [Asterisk-Users] Re: RE : Re: RE : codecs order and so on
Ok Oliver, i have been messing arund with this codec thing, and this may help you as it made it with my problem: http://bugs.digium.com/view.php?id=4825 I have adapted a patch from the one posted there, so it can apply in asterisk-1.2.1, you can get the patch here (not sure how much time will i leave it there): http://chewbacca.ivsol.net/codec_negotiation-20051107-ivsol0.patch Unfortunatly i have customized that patch taking the sources from gentoo ebuild package, and may not apply cleanly in virgin sources. Regards. On 1/12/06, Olivier Taylor [EMAIL PROTECTED] wrote: Thanks for all, But Asterisk is able to use g729 pass-tru and both ends have g729, then the question is: Why asterisk doesn't use g729 pass-thru when both ends have it? For incoming calls from Voip, G729 is not a problem, problems appears when I make a call to Voip... Olivier Ps: No need to answer, that's just a fact -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tomislav Parcina Envoyé : jeudi 12 janvier 2006 10:31 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Re: RE : Re: RE : codecs order and so on In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Calling zap = no problem, Ulaw is choosen Calling pstn provider =fail (I need g729 but Ulaw is choosen) Call from zap = no problem Ulaw is choosen Call from pstn = no problem g729 used... When you call out * establishes two channels. One is between Ua and *, and another between * and Zap (or provider). If you call out, asterisk first negotiate codec for that channel. Then it tries to nagotiate codec for second channel. When you call your provider it can't nagotiate because he doesn't have g729 codec. This is reason why you have problem, and I have explain how to solw it. There is nothing else I can say to help you. -- Tomislav Parcina [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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] dCAp
You should have no problem with the practical, however the written part of the exam does have a lot of standards (IETF) questions and the like. If you are very knowledgeable in SIP and other VoIP standards and are familiar with the config files you may do well. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of blackgeckoSent: Thursday, January 12, 2006 9:26 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] dCAp HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk´s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and outbound routing, VoIP providers like voipjet, etc. Do you think the bootcamp is a good option???thanks ___ --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] Zaptel issues
On a side note: When poking around, I noticed in the zaptel Makefile that there is a section talking about ztdummy automatically being included on 2.6 kernels. Is this correct? On to the main topic: Any ideas for troubleshooting this? [EMAIL PROTECTED] zaptel-1.2.1]# /etc/rc.d/init.d/zaptel startLoading zaptel framework: FATAL: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format [FAILED]Waiting for zap to come online...Error: missing /dev/zap! [EMAIL PROTECTED] libpri-1.2.1]# modprobe ztdummyWARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module formatWARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module formatFATAL: Error inserting ztdummy (/lib/modules/2.6.8-022stab050.1/extra/ztdummy.ko): Invalid module formatFATAL: Error running install command for ztdummy Mike HammettIntelligent Computing Solutionshttp://www.ics-il.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] app_rxfax.so and app_txfax.so
Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading module app_rxfax.so failed! Ouch ... error while writing audio data: : Broken pipe Anyone can help me ? Thanks Giordano ___ --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] dCAp
I second this.. I have been working with asterisk for over a year now and when I took it at astricon 2005 it was very challenging. Put it this way Mark Spencer failed the test. Its not something you want to walk into blind. The hands on portion is cake. However the the written exam is very challenging. Like BJ Said. You need to be well rounded in asterisk. brian On Thu, 2006-01-12 at 09:54 -0500, BJ Weschke wrote: On 1/12/06, blackgecko [EMAIL PROTECTED] wrote: HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk´s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and outbound routing, VoIP providers like voipjet, etc. Do you think the bootcamp is a good option??? As someone that took the dCAP exam, unless you're extremely comfortable with ALL aspects of Asterisk, I'd go for the Bootcamp. The dCAP written and practical covers pretty much everything about Asterisk. I was already doing some development with Asterisk and still found it challenging when I took it, so I'd recommend the bootcamp if you don't want to keep paying to retake the test. -- Bird's The Word Technologies, Inc. http://www.btwtech.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 -- _.._ Brian Fertig Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --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 and Radius
http://www.voip-info.org/wiki-PortaOne+Radius+auth Have you already checked this? not working? regards On 1/11/06, Wile [EMAIL PROTECTED] wrote: I need to integrate Asterisk with radius server, because I need auth and signaling from radius (i.e start and stop messages in realtime) (or something that do that). I can't find an implementation of this with enough documentation to install and run.- Wile ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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: Failover Device?
On 1/12/06, Tomislav Parcina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. I'm sure that I'm not the only person that has notice that there is lots of people that start new thread by replaying to old message. That way neither them, or lots of other people, sees that mail as new therad. Yeah I've noticed that too.. I don't do that though. Ok on to the question at hand. I am trying to fail over asterisk. I have PRI redundancy. What I need, however, is someway to transfer the PRI from asterisk box A to asterisk box B if asterisk box A fails. So while, yes, I can build a second asterisk box and use SER, or DNS or whatever to point my sip devices to it... the question is how do I get the PRIs to know which box to route to? ___ --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] Transfer issue with a Cisco CCM/phone
Hello, We have a mixed environment here consisting of a number of Avaya PBX systems, a group of Cisco Call Managers, an H.323 gateway on a Cisco router, and an Asterisk server. The PBX land is connected to the VoIP land using the Cisco router/H.323 gateway. The Asterisk system is running code from the CVS tree from around mid Oct 2005.The OH323 driver on the system is from inaccessnetworks, version 0.7.3. We are having an issue transferring calls from one system to another. When the transfer is attempted on various sets, it does not work and the call is 'lost'. When the called party is on a cisco phone, the Avaya (calling party) hears music on hold from the cisco system while the transfer is being made. The Cisco user reaches the auto-attendant on the Asterisk box and the music on hold is heard by the Avaya caller. When the Cisco user attempts to complete the transfer by pressing the transfer button again, the music goes away on the Avaya phone but the call remains on the Cisco phone and the transfer button remains active on the cisco set and does not transfer the call. Additional examples: THESE DO NOT WORK... * Avaya user calls a Cisco user and talks. The called party (Cisco user) then attempts to transfer to an auto-attendant on the Asterisk system. The call does not transfer. * Cisco user calls another Cisco user and talks. The called party then attempts to transfer to an auto-attendant on the Asterisk system. The call does not transfer. * Outside user calls number which passes through Avaya PBX and Cisco router to the Auto attendant. The user then dials an extension to a Cisco phone. The called party on the Cisco phone can not transfer a call. THESE WORK... * Avaya user calls another Avaya user and talks. The called party then attempts to transfers to an auto-attendant on the Asterisk system. The call DOES TRANSFER. * Cisco user calls an Avaya user and talks. The called party (Avaya) then attempts to transfer to an auto-attendant on the Asterisk system. The call DOES TRANSFER. * Outside user calls number which passes through Avaya PBX and Cisco router to the Auto attendant. The user then dials an extension to an Avaya phone. THE CALLED PARTY CAN TRANSFER THE CALL. Any assistance in solving this issue would be appreciated. [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] read .what else to do ?
Please note that recent IOS has SIP NAT traversal turned on by default. I believe that it only supports internal UA / external server. Since you also want the opposite, you should probably turn it off: no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060 Some IOS versions will even crash on SIP behind NAT. See http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. You can edit rtp.conf to use e.g 1-10007 (would allow 4 calls) and then only 8 ip nat statements would be needed for RTP. You don't say what's failing. make calls outside our LAN sounds like you are trying to call using a VoIP provider that Asterisk registers with. But your remote SIP phones is something different; which of the above are failing? Are the registrations successful? Is it just the RTP that's not working (in which case the called phone will still ring)? If not, what error or timeout is reported? If * verbose and/or debug logs don't show precisely what is going wrong, use Ethereal (on both sides of the router if necessary) to see what is happening. --Stewart Hi all , I have tried configuring Asterisk at home to make calls outside our Lan WITHOUT any success (Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ) To be precise i did the following (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 1 to 2 to 192.168.1.2 (2) I set externip = x.x.x.x (to our public WAN) localnet =192.168.1.0 /255.255.255.0 (3) I also set nat=yes qualify=yes (4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ).Please advise me how to make it work !!! (5) I am using xten lite soft phone on my pc . (6) I use cisco 1700 series router ,and i have natting configured on this router .Maybe I am using a wrong command .Please,tell me the commands to forward the ports Port 5060-5082,1 to 2 to 192.168.1.2 on a cisco router . Please reply and advice !!! Thanks ___ --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] app_rxfax.so and app_txfax.so
0.0.3 series releases are for development only. Roll back to 0.0.2-pre21 and you should be good. hth -Original Message-From: Giordano Grandis [mailto:[EMAIL PROTECTED]Sent: Thursday, January 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] app_rxfax.so and app_txfax.so Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading module app_rxfax.so failed! Ouch ... error while writing audio data: : Broken pipe Anyone can help me ? Thanks Giordano ___ --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] Adit 600 and echo
I'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around with the TX/RX gains on the channel bank will not do anything and this needs to be resolved via the TX gains within Asterisk? I was able to set the TXGAIN to -6.3 and it did help some, but if I try -6.4 or more on any of the channels, I can no longer here audio when calling into the facility. Any suggestions? Doug ___ --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] app_rxfax.so and app_txfax.so
I wonder if there might be a reason why the download directory for spandsp has a notice saying: SPANDSP_0.0.2_IS_FOR_USERS SPANDSP_0.0.3_IS_FOR_DEVELOPERS Steve Giordano Grandis wrote: Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading module app_rxfax.so failed! Ouch ... error while writing audio data: : Broken pipe Anyone can help me ? Thanks **Giordano** ___ --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] DTMF Issues With Asterisk 1.2 IVR
Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that requires you to enter into an ivr system. I already set my dtmf mode in asterisk. ___ --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] Company directory not finding names... sometimes.
I am using Asterisk 1.2 and have had several complaints of people calling in and using the company directory to find an extension by first name and being told that no entry is found for the name dialed (they are definitely dialing the proper name). However I try it and others have tried it and it works fine. Has anyone else experienced this? Any suggestions on how I can diagnose/fix this? Thanks! ___ --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 to register a SIP phone on Asterisk behind NAT
Hi everybody, One of my client's Asterisk box is behind NAT. They have only one public IP on which they have their router. I can access the Asterisk server using port forwarding (port 22) for SSH. Now this client wants to connect two SIP phones to this Asterisk box from two remote locations. How can this be done. If I forward ports, e.g. 5060-5070 to this Asterisk box, there is no guarantee that the SIP phones will be using the same ports from the remote locations, because ports get changed over the Internet, like for another scenario with my client on public IP, remote SIP ports are 17355, 61949, 61666 etc., though they are configured 5060 on the phones, and 5060 in sip.conf. What is the solution in this scenario of registering SIP on Asterisk behind NAT? Thanks, Zeeshan A Zakaria ___ --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] Company directory not finding names... sometimes.
David C. Nicosia wrote: I am using Asterisk 1.2 and have had several complaints of people calling in and using the company directory to find an extension by first name and being told that no entry is found for the name dialed (they are definitely dialing the proper name). However I try it and others have tried it and it works fine. I believe it goes against last name. Doug ___ --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] Problem with an automatic responder
Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time ago). I can post pri debug output in both cases, if needed. Thanks in advance for any help -- Mimmus ___ --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
R: [Asterisk-Users] app_rxfax.so and app_txfax.so
I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ? Thanks Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Colin Anderson Inviato: giovedì 12 gennaio 2006 17.20 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] app_rxfax.so and app_txfax.so 0.0.3 series releases are for development only. Roll back to 0.0.2-pre21 and you should be good. hth -Original Message- From: Giordano Grandis [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] app_rxfax.so and app_txfax.so Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading module app_rxfax.so failed! Ouch ... error while writing audio data: : Broken pipe Anyone can help me ? Thanks Giordano ___ --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 crossed lines?
Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office are automatically connected with an incoming line. Anyone seen this or know what might be causing it? ___ --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] PBX making ENUM lookups
Hello
I have a Siemens HiPath and I wanted to make him do ENUM lookups.
Then I connected it to an Asterisk (with ISDN) and route all calls to
Asterisk.
Then, Asterisk does the ENUM lookup, this way:
exten= _XXX,1,BackGround(nic.at/enum-doing)
exten= _XXX,2,EnumLookup(351${EXTEN:})
exten= _XXX,3,BackGround(nic.at/enum-successful)
exten= _XXX,4,Dial(${ENUM},30,r)
But how do I configure Asterisk to deliver the call back to the
Siemens PBX, if he doesnt find an ENUM match or if the contact is offline?
(I need it because its the Siemens PBX thats connected to the PSTN)
Thanks
Joao Pereira
___
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Re: [Asterisk-Users] Asterisk and Radius
El jue, 12 de ene de 2006, a las 09:55:06 -0600, Moises Silva dijo: http://www.voip-info.org/wiki-PortaOne+Radius+auth Have you already checked this? not working? regards I'm trying to ^^ this working but I can't do yet. :( Wile ___ --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] app_rxfax.so and app_txfax.so
Best practice is to delete app_rxfax/txfax, delete libspandsp* from /usr/local/lib, recompile SpanDSP,and recompile Asterisk then you know for sure you have a good install. In /usr/local/lib a link is created to link libspandsp.so to the actual library and if you don't explicitly delete the links and the library when you reinstall you run into problems. I'm sure Mr Underwood will roll his eyes when he sees this, but doing it this way makes it un-ambiguous for both the system and you. -Original Message-From: Giordano Grandis [mailto:[EMAIL PROTECTED]Sent: Thursday, January 12, 2006 9:58 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: R: [Asterisk-Users] app_rxfax.so and app_txfax.so I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ? Thanks Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Colin AndersonInviato: giovedì 12 gennaio 2006 17.20A: 'Asterisk Users Mailing List - Non-Commercial Discussion'Oggetto: RE: [Asterisk-Users] app_rxfax.so and app_txfax.so 0.0.3 series releases are for development only. Roll back to 0.0.2-pre21 and you should be good. hth -Original Message-From: Giordano Grandis [mailto:[EMAIL PROTECTED]Sent: Thursday, January 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] app_rxfax.so and app_txfax.so Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading module app_rxfax.so failed! Ouch ... error while writing audio data: : Broken pipe Anyone can help me ? Thanks Giordano ___ --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] Where do I find *asterisk-capi*
Pls, where do I find asterisk-capi I am using now asterisk 1.2.1 with a SuSE 9.3 in SuSE 9.3 there was the old version for 1.0.6 ... can I use that old asterisk-capi for the current and on my system installed version 1.2.1 ??? thnx chris ___ --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] cisco as5400, sip, asterisk. cisco won't detect that the call is answered
We've got this configuration : Cisco as5400 --- asterisk main server asterisk for cells gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind the cisco talks the one after the gsm gateway will hear him, but not the contrary. (like when you have a problem with nat, plus the I'm still hearing the ringingtone problem) ((no, cisco is on a public IP, also the two asterisk servers, and all sip is canreinvite=no) the dial chain is something like : asterisk main server: [cisco context] X.,1,Dial(iax/[EMAIL PROTECTED] for cells) asterisk for cells: [cisco context] X.,1,Dial(sip/[EMAIL PROTECTED] gateway) If the main server dialplan becomes like : [cisco context] X.,1,Answer X.,n,Dial(iax/[EMAIL PROTECTED] for cells) the problem is solved, but all the calls are seen as answered by the cisco (well, they are) and this is not good for billing purposes. (the 'asterisk for cell' server writes the cdr duration / billsec correctly, but trust is not of the business world ) (there are some lynksis paps connected to the asterisk main server, and they work perfectly) ___ --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] Problem with an automatic responder
I would be useful if you could post your config files and the pri debug as well. check your zapata.conf or paste it here so we can take a look.AlyedReturn-Path: [EMAIL PROTECTED] Thu Jan 12 10:04:28 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 12 Jan 2006 10:04:28 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6DAB2C258; Thu, 12 Jan 2006 09:58:40 -0700 (MST)Received: from psmtp.com (exprod5mx128.postini.com [64.18.0.42]) by lists.digium.com (Postfix) with SMTP id 4101AC24B for asterisk-users@lists.digium.com; Thu, 12 Jan 2006 09:58:34 -0700 (MST)Received: from source ([151.9.129.69]) (using TLSv1) by exprod5mx128.postini.com ([64.18.4.10]) with SMTP; Thu, 12 Jan 2006 10:58:41 CSTReceived: from aulin (aulin.pitagora.it [193.227.67.249]) by allserv.pitagora.it (8.12.11/8.12.11) with ESMTP id k0CGwXKd006897 for asterisk-users@lists.digium.com; Thu, 12 Jan 2006 17:58:33 +0100Received: from VIGGIANI ([193.227.66.207]) by MERCURIO.pitagora.it with Microsoft SMTPSVC(6.0.3790.1830); Thu, 12 Jan 2006 17:58:32 +0100X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comFrom: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Thu, 12 Jan 2006 17:58:32 +0100Message-ID: [EMAIL PROTECTED]MIME-Version: 1.0Content-Type: text/plain; charset="us-ascii"Content-Transfer-Encoding: 7bitX-Mailer: Microsoft Office Outlook 11Thread-Index: AcYXmWqjqLUMCcjoQT2Cz9B43oUplQ==X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2180X-OriginalArrivalTime: 12 Jan 2006 16:58:32.0672 (UTC) FILETIME=[6B723600:01C61799]X-Spam-Status: No, score=-4.8 required=5.5 tests=ALL_TRUSTED,AWL,BAYES_00 autolearn=ham version=3.0.4X-Spam-Checker-Version: SpamAssassin 3.0.4 (2005-06-05) on allserv.pitagora.itX-Virus-Scanned: ClamAV version 0.88, clamav-milter version 0.87 on localhostX-Virus-Status: CleanX-pstn-levels: (S: 4.16697/99.78420 )X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1X-pstn-addresses: from [EMAIL PROTECTED] [90/4]Subject: [Asterisk-Users] Problem with an automatic responderX-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listReply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comList-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.comList-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED]List-Archive: http://lists.digium.com/pipermail/asterisk-usersList-Post: mailto:asterisk-users@lists.digium.comList-Help: mailto:[EMAIL PROTECTED]List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED]Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: [EMAIL PROTECTED]Hi,I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phonesand a few of VoIP phones directly connected to Asterisk.Calling a number (only one until now!) - an automatic responder (IVR) - fromVoIP phones works, from analog phones doesn't work: NOANSWER after a fewseconds. I'm using no 'r' in dial options (this caused a problem with an IVRsome time ago).I can post pri debug output in both cases, if needed.Thanks in advance for any help-- Mimmus___--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
[Asterisk-Users] (Trunk) in production
Just out of curiosity, how many of you are using trunk in a production environment? Are you performing regular compilations of the code as well? Do you explicitly prefer trunk over stable, or vice versa?Ronald Lewis www.ronaldlewis.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] AMP and additional conf files
Title: Message
Hello
all. I've been searching and can't quite find what I'm looking
for...
I've
gotten AMP installed and up and running quite decentlyonan Asterisk box and am now in the
process of tweaking it to my needs. My company currently has around 70
employees and we are running on a complete Avaya system, but this system is no
longer going to work for us (too much money for not enough stuff). So I
have been put in charge of setting up an Asterisk PBX and get an entire test
system going on it to see if Asterisk will
meet ourtelephone needs. Extensions, queues, voicemail,
stats, etc etc. Here's the problem:this Asterisk server is actually
currently running live, serving information to people calling in to it. I
need my test office setup, with AMP and this other system to work
simultaneously, but yet totally separate. As my stuff is for a test, I
would like to set it up so that when I dial inTO myAsterisk
PBXFROM a specific telephone number, it takes me to my office test section
in asterisk, otherwise, from ANY other number, it dials the info serving
section. This would allow me to call froma certain telephone number
andbe able to get to my test office setup, but if anybody else calls from
any other number, they get theother stuff. Doesn't sound too bad
right?
So how
would one do this using AMP if AMP is more of the "secondary" system? If I
understand correctly, to add additional, custom contexts to extensions.conf, it
should be entered into extensions_additional.conf and the contexts should
contain the word "custom" in them. So, first question, what if I want that
custom context to be the first context (as in possibly the default context), but
only if it's from a certain telephone number...? I assume you
wouldenter that custom context as thecontext inzapata.conf,
but how would you tell it to go back to the AMP stuff if the FROM telephone
number is my speicifc telephone number? What context would I send it to so
that it will do the regular AMP stuff? (Incidentally, I have a local
telephone number and an 888 telephone number coming into my PRI, but when
called, my Asterisk PBXviews/receives them both as the local telephone
number.)
Also,
what if my custom contexts in extensions_additional.conf call macros? Do
my macro names need to contain the word "custom" as well?
Where
to put my nineoneone context for it to utilize the defined
globals?
Belowis my currentextensions.conf file.(My
extensions_additional.conf is currently empty.) The first part is
the custom stuff that I use to serve up the info when customers call in. I believe
this is what should be moved to the extensions_additional.conf... I didn't
post the entire extensions.conf file as the rest is the standard stuff set up by
AMP. Any suggestions?
Thanks,
Ben
Ferguson
[general]static=yeswriteprotect=yes
IAXINFO=guest
; IAXtel
username/passwordTRUNK=Zap/g1
; Trunk
interfaceTRUNKMSD=1
; MSD digits to strip (usually 1 or 0)
[globals]
EMERGENCY=0EMERGENCY_TRUNK=Zap/17; Change this for production
use:EMERGENCY_NUM=911
[nineoneone]exten = s,1,SetVar(SET_EMERG_FLAG=0)exten =
s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})exten =
s,n,SetGlobalVar(EMERGENCY=1)exten =
s,n,SetVar(SET_EMERG_FLAG=1)exten =
s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})exten =
s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)exten =
s,n,SoftHangup(${EMERGENCY_TRUNK}-1)exten = s,n,Wait(12)exten =
s,n,Goto(checkavail)exten = s,s+2(inprogress),Congestionexten =
s,checkavail+101(notavail),Goto(trunkbusy)exten =
h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?3)exten =
h,3,SetGlobalVar(EMERGENCY=0)exten = t,1,goto(s,1)
[local];; Master context
for local, toll-free, and iaxtel calls only;ignorepat = 9include
= default
[default]exten =
s,1,Answerexten = s,2,Wait(1)exten =
s,3,Background(cc_welcome)
;if someone enters a tour
numberexten = _XX,1,AGI(getAudioFile.php,${EXTEN})exten =
_*XX,1,AGI(loanOfficerLogin.php,${EXTEN})
;if someone enters an invalid
extensionexten = i,1,Playback(cc_sorry)
;timeoutexten =
t,1,goto(s,2)
;incoming numbersexten
=551212,1,goto(s,1)exten
= 8885551212,1,goto(s,1)
;*;***Message
Sections;*
[macro-setPinVars]exten =
s,1,SetGlobalVar(LAST_AUDIO=${ARG1})exten =
s,2,SetGlobalVar(AGENT_PIN=${ARG2})exten =
s,3,SetGlobalVar(HAS_AUDIO=${ARG3})exten =
s,4,goto(getSponserPin,s,1)
[getSponserPin]exten =
s,1,Background(agent-pass);exten = s,1,SayDigits(${AGENT_PIN})exten
= ${AGENT_PIN}#,1,GotoIf($["${HAS_AUDIO}" = "yes"]?3:2)exten =
${AGENT_PIN}#,2,goto(loRecordMessage,s,1)exten =
${AGENT_PIN}#,3,goto(loHasMessage,s,1)
exten = i,1,Wait(1)exten
= i,2,Playback(cc_sorry)
[loRecordMessage]exten =
s,1,Background(to-compose-a-message)exten =
s,2,Background(press-1)exten =
s,3,Background(T-to-rtrn-to-main-menu)exten =
s,4,Background(press-9)
exten =
1,1,Playback(vm-intro)exten =
Re: [Asterisk-Users] AMP and additional conf files
On Thu, January 12, 2006 19:18, Ben Ferguson said: Hello all. I've been searching and can't quite find what I'm looking for... I've gotten AMP installed and up and running quite decently on an Asterisk box and am now in the process of tweaking it to my needs. My company currently has around 70 employees and we are running on a complete Avaya system, but this system is no longer going to work for us (too much money for not enough stuff). So I have been put in charge of setting up an Asterisk PBX and get an entire test system going on it to see if Asterisk will meet our telephone needs. Extensions, queues, voicemail, stats, etc etc. Here's the problem: this Asterisk server is actually currently running live, serving information to people calling in to it. I need my test office setup, with AMP and this other system to work simultaneously, but yet totally separate. As my stuff is for a test, I would like to set it up so that when I dial in TO my Asterisk PBX FROM a specific telephone number, it takes me to my office test section in asterisk, otherwise, from ANY other number, it dials the info serving section. This would allow me to call from a certain telephone number and be able to get to my test office setup, but if anybody else calls from any other number, they get the other stuff. Doesn't sound too bad right? So how would one do this using AMP if AMP is more of the secondary system? If I understand correctly, to add additional, custom contexts to extensions.conf, it should be entered into extensions_additional.conf and the contexts should contain the word custom in them. So, first question, what if I want that custom context to be the first context (as in possibly the default context), but only if it's from a certain telephone number...? I assume you would enter that custom context as the context in zapata.conf, but how would you tell it to go back to the AMP stuff if the FROM telephone number is my speicifc telephone number? What context would I send it to so that it will do the regular AMP stuff? (Incidentally, I have a local telephone number and an 888 telephone number coming into my PRI, but when called, my Asterisk PBX views/receives them both as the local telephone number.) SNIP Normally in AMP (depending on version) you'd make either an inbound route like this : 4081234567|4081234599 (where the 4567 is the DID and 4599 the callerID) or an inbound route with DID=4081234567 and CID=4081234599 and then send it to a specific extension or custom context... HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Zaptel SVN
On Jan 12, 2006, at 8:33 AM, Kevin P. Fleming wrote: BJ Weschke wrote: It has just been fixed in /trunk That fix should be merged over to branch-1.2 as well then, if needed. It shouldn't be; it's something left over from a merge into HEAD that I made the other day. Matthew Fredrickson ___ --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] Using an extension to send a linux command
I am a newbie to asterisk and am trying to send a linux command using extensions in asterisk, for example when I dial I want to run the linux command /usr/local/bin/br -c -n 1 (obviously without the quotes). If I SSH into my asterisk box and enter that command, it works, however I can't seem to get it to work from asterisk. I am running [EMAIL PROTECTED] (I know, I am a sucker for a GUI). Below is what I have in my dialplan. Watching the CLI output it seems to be running the priorities correctly, and even assumedly sending the command, however the script (br) never is actually executed. Any ideas? (no I don't want to convert the script into an agi or php yet, I like it as it is) exten = ,1,Goto(custom-command,s,1) [custom-command] exten = s,1,System(/usr/local/bin/br -c C -n 1) exten = s,n,Hangup() Kaleb L. Kunzler ___ --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] Using an extension to send a linux command
What user is yoru asterisk service running as? It is probably a permissions or path issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kaleb L. Kunzler Sent: Thursday, January 12, 2006 1:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Using an extension to send a linux command I am a newbie to asterisk and am trying to send a linux command using extensions in asterisk, for example when I dial I want to run the linux command /usr/local/bin/br -c -n 1 (obviously without the quotes). If I SSH into my asterisk box and enter that command, it works, however I can't seem to get it to work from asterisk. I am running [EMAIL PROTECTED] (I know, I am a sucker for a GUI). Below is what I have in my dialplan. Watching the CLI output it seems to be running the priorities correctly, and even assumedly sending the command, however the script (br) never is actually executed. Any ideas? (no I don't want to convert the script into an agi or php yet, I like it as it is) exten = ,1,Goto(custom-command,s,1) [custom-command] exten = s,1,System(/usr/local/bin/br -c C -n 1) exten = s,n,Hangup() Kaleb L. Kunzler ___ --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] read .what else to do ?
Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. just add the range ports tih a ":" e.g 192.168.1.2 1 : 10007(4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ).Please advise me how to make it work !!! If what you are trying to do is a SIP -- NAT -- Internet -- Nat -- Asterisk call them I'm afraid you would need to use a SIP/RTP router. Alyed Return-Path: [EMAIL PROTECTED] Thu Jan 12 09:29:42 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Please note that recent IOS has SIP NAT traversal turned on by default.I believe that it only supports internal UA / external server.Since you also want the opposite, you should probably turn it off:no ip nat service sip tcp port 5060no ip nat service sip udp port 5060Some IOS versions will even crash on SIP behind NAT. Seehttp://lists.digium.com/pipermail/asterisk-users/2004-January/033718.htmlSorry, I don't know how to forward a range of ports. To forwarda single port, use something like:ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendablewhere x.x.x.x is your public IP.You can edit rtp.conf to use e.g 1-10007 (would allow 4 calls) andthen only 8 ip nat statements would be needed for RTP.You don't say what's failing. "make calls outside our LAN" sounds likeyou are trying to call using a VoIP provider that Asterisk registerswith. But "your remote SIP phones" is something different; which ofthe above are failing? Are the registrations successful? Is it justthe RTP that's not working (in which case the called phone will stillring)? If not, what error or timeout is reported?If * verbose and/or debug logs don't show precisely what is going wrong,use Ethereal (on both sides of the router if necessary) to see whatis happening.--Stewart Hi all , I have tried configuring Asterisk at home to make calls outside our Lan WITHOUT any success (Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ) To be precise i did the following (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 1 to 2 to 192.168.1.2 (2) I set externip = x.x.x.x (to our public WAN) localnet =192.168.1.0 /255.255.255.0 (3) I also set nat=yes qualify=yes (4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ).Please advise me how to make it work !!! (5) I am using xten lite soft phone on my pc . (6) I use cisco 1700 series router ,and i have natting configured on this router .Maybe I am using a wrong command .Please,tell me the commands to forward the ports Port 5060-5082,1 to 2 to 192.168.1.2 on a cisco router . Please reply and advice !!! Thanks___--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] How to register a SIP phone on Asterisk behind NAT
I currently do this for about 30 different cisco 79xx's connecting to some hosted Asterisk servers. Asterisk listens by default for any SIP connection on UDP port 5060. And will use RTP UDP port 1 to 2 The phones use UDP Port 5061 for incoming connections (from Asterisks or other SIP Devices) and use for RTP, UDP port 1 to 2. Now, if you are going to have the two remote phones at two separate locations then you can have the firewall forward these ports to the IP Address of the SIP Phonesnot we need to discuss how do you over come NATing. I use Cisco phones...so I setup the external IP Address (the address that the remote phone will appear as) in the Configuration and Turn on NATing. This makes the phone use this address in the SIP communications. Asterisks has no idea that the phone is being NATed and has NAT turned off. If you have the phones at the same location, then you need to configure the phone to use different ports for both SIP Communications and RTP. I use 5061 for the first phone and then go up from there, 5062 for second phone. I then use 1 to 11999 for RTP for the first phone and 12000 to 13999 for the second phone. The cisco config allows me to enter these values also in the configuration. If you are using some other phones, the you will need to figure out how to configure them to do the same...basically the phones will send out packets with the Internet Routable addresses and the port info configured for them. Kevin J. Steil Steil Technologies -Original Message- From: Zeeshan [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 11:46 AM To: Asterisk User List Subject: [Asterisk-Users] How to register a SIP phone on Asterisk behind NAT Hi everybody, One of my client's Asterisk box is behind NAT. They have only one public IP on which they have their router. I can access the Asterisk server using port forwarding (port 22) for SSH. Now this client wants to connect two SIP phones to this Asterisk box from two remote locations. How can this be done. If I forward ports, e.g. 5060-5070 to this Asterisk box, there is no guarantee that the SIP phones will be using the same ports from the remote locations, because ports get changed over the Internet, like for another scenario with my client on public IP, remote SIP ports are 17355, 61949, 61666 etc., though they are configured 5060 on the phones, and 5060 in sip.conf. What is the solution in this scenario of registering SIP on Asterisk behind NAT? Thanks, Zeeshan A Zakaria ___ --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] Using an extension to send a linux command
It is running as asterisk ([EMAIL PROTECTED] default). I have tried 'chown asterisk:asterisk br' as well as 'chmod 775 br' but nothing seems to help. Kaleb -Original Message- From: Alexander Lopez Sent: Thursday, January 12, 2006 11:59 AM Subject: RE: [Asterisk-Users] Using an extension to send a linux command What user is yoru asterisk service running as? It is probably a permissions or path issue. ___ --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] MTU and Voice Delay (latency??)
Rich Adamson wrote: What happens if you set the 3com AND Speedtouch to full duplex? Setting both to Full Duplex (10 or 100) shuts the port on the 3COM switch down! ___ --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] No D-channels available! Using Primary on channel 16 anyway!
I had a very similar problem some months ago, was using a Sangoma A101 card though. The problem was something related to the card's memory and was able to solve it by updating the driver. It was caused due to I was using a brand new card with a not so updated driver (I was using one that I thought was "stable") So my advice here is to check the driver version you are using if not the very last one, then update it. Try looking at the /var/log/messages file for any extra info, you might find something interesting. Alyed Return-Path: [EMAIL PROTECTED] Thu Jan 12 06:16:56 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 12 Jan 2006 06:16:56 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6834A4066; Thu, 12 Jan 2006 06:15:45 -0700 (MST)Received: from psmtp.com (exprod5mx148.postini.com [64.18.0.180]) by lists.digium.com (Postfix) with SMTP id ACB784060 for asterisk-users@lists.digium.com;Hi!I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and * 1.2.Every few hours I get this message and asterisk dies just after that:Warning: No D-channels available! Using Primary on channel 16 anyway!When this happens restarting zaptel and asterisk services, generally puts the system back onlinemy zaptel.con reads:span=1,1,0,ccs,hdb3 #span=2,0,0,cas,hdb3 span=3,2,0,esf,b8zs #-- This because we have two American CBsspan=4,3,0,esf,b8zsbchan=1-15dchan=16bchan=17-31fxoks=63-86fxoks=87-110loadzone = usIdeas anybody? Please? Things done:* zttool/ztcfg* Trying R2 instead of PRI (R2 is the south americanstatdar, which wont even start)*Added crc4 to span1, with ugly sound consequences-- Paavum Regina, Per Secula et Secularum!! ___ --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] Using an extension to send a linux command
You may be having an issue with the arguments. Try 'wraping the script within another (ie runbr) Also login as asterisk user, you may need to change /etc/passwd to give asterisk a shell, I am not sure as I don't do @ Home. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kaleb L. Kunzler Sent: Thursday, January 12, 2006 2:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Using an extension to send a linux command It is running as asterisk ([EMAIL PROTECTED] default). I have tried 'chown asterisk:asterisk br' as well as 'chmod 775 br' but nothing seems to help. Kaleb -Original Message- From: Alexander Lopez Sent: Thursday, January 12, 2006 11:59 AM Subject: RE: [Asterisk-Users] Using an extension to send a linux command What user is yoru asterisk service running as? It is probably a permissions or path issue. ___ --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] Re: Transfer issue with a Cisco CCM/phone (Peckham, Christopher)
Christopher-Nothing like defining a complicated environment. I do have some experience in this arena- but unfortunately, not with the OH323 driver- I generally stick to the Nufone driver, as I find it more reliable overall. YMMV. One thing that might help is if you could tell us if it ever worked, or if this is a new problem that's cropped up since a particular change. Still- there are two areas to check- one, I'd start up some debugs on your gatekeeper, to see if the call is being signalled properly to Asterisk, and on Asterisk itself, to see what's being passed in. This is critical- I'm betting this is a simple case of dialplan mangling- but only the debug logs will tell. Secondly, I know that in the CCM trunk definitions, it's important to ensure the trunk definition follows the recommendations *exactly*. A big killer here is the 'Media Termination Point Required' box- generally, for transfers, you need one- and it needs to be functional. Your CCM administrator should be able to verify that it's defined and working- but CCM is not a simple setup. (I've done it, a couple of times). There are also version differences- you don't mention what release of CCM you're running, for example. CCM doesn't always comply with 'the rules' of H.323 as we know them- they are known to do some non-standard things, and various channel drivers can take offence with that. If SIP is available (CCM 4.0 and above), you may want to consider re-architecting to it, as a good many of these problems go away under SIP. Let me know if I can be of further assistance.-Paul DavidsonPlanCommunications, LLC.On 1/12/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Message: 4Date: Thu, 12 Jan 2006 11:01:02 -0500From: Peckham, Christopher [EMAIL PROTECTED]Subject: [Asterisk-Users] Transfer issue with a Cisco CCM/phone To: asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCIIHello,We have a mixed environment here consisting of a number of Avaya PBXsystems, a group of Cisco Call Managers, an H.323 gateway on a Cisco router, and an Asterisk server.The PBX land is connected to the VoIPland using the Cisco router/H.323 gateway.The Asterisk system is running code from the CVS tree from around midOct 2005.The OH323 driver on the system is from inaccessnetworks, version 0.7.3.We are having an issue transferring calls from one system to another.When the transfer is attempted on various sets, it does not work and thecall is 'lost'. When the called party is on a cisco phone, the Avaya (calling party) hears music on hold from the cisco system while thetransfer is being made.The Cisco user reaches the auto-attendant onthe Asterisk box and the music on hold is heard by the Avaya caller.When the Cisco user attempts to complete the transfer by pressing the transfer button again, the music goes away on the Avaya phone but thecall remains on the Cisco phone and the transfer button remains activeon the cisco set and does not transfer the call.Additional examples: THESE DO NOT WORK...* Avaya user calls a Cisco user and talks. The called party (Cisco user)then attempts to transfer to an auto-attendant on the Asterisk system.The call does not transfer. * Cisco user calls another Cisco user and talks.The called party thenattempts to transfer to an auto-attendant on the Asterisk system. Thecall does not transfer.* Outside user calls number which passes through Avaya PBX and Cisco router to the Auto attendant.The user then dials an extension to aCisco phone.The called party on the Cisco phone can not transfer acall.THESE WORK...* Avaya user calls another Avaya user and talks.The called party then attempts to transfers to an auto-attendant on the Asterisk system.Thecall DOES TRANSFER.* Cisco user calls an Avaya user and talks.The called party (Avaya)then attempts to transfer to an auto-attendant on the Asterisk system. The call DOES TRANSFER.* Outside user calls number which passes through Avaya PBX and Ciscorouter to the Auto attendant.The user then dials an extension to anAvaya phone.THE CALLED PARTY CAN TRANSFER THE CALL. Any assistance in solving this issue would be appreciated.[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] Second edition of my * book has been release d
But for us? From: William Boehlke [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 2:24 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Second edition of my * book has been released $39.95 retail. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel SartorSent: Tuesday, January 10, 2006 6:27 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Second edition of my * book has been released How much is this book ?? 2006/1/10, Randy Williams [EMAIL PROTECTED]: Greetings All,I have found that Paul's book is just right for rounding out the edgeswhen getting started.I managed to temporarily migrate our T-1 Asterisk system to a Analog asterisk system on information in Paul's book alone.Nicely done and a neat bit of help in a pinch.Just my $0.02 USD you understand. :)RandyWPaul Mahler wrote:Hi Greg, My book is a good place for a beginner to get started. I also find it to beuseful as a reference for Asterisk. It's not an advanced book, there areadvanced features it doesn't cover, for example AGI or the management interface.It should be very helpful for your customers. It should be helpful for abeginning to intermediate administrator. I still frequently refer to itmyself when I'm having a senior moment. :) There isn't anything in the book that would make it less useful for the CVSor stable branches.The O'Reilly book is excellent. I think my book complements the O'Reillybook. If I were just starting I would buy both. I think my book may be a bit more useful as a reference. I think I cover a bit more beginner's territory.Hope This Helps,Paul-Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] ] On Behalf Of [EMAIL PROTECTED]Sent: Monday, January 09, 2006 9:10 PMTo: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Second edition of my * book has beenreleasedHow does it compare with the O'Rielly book?Does it include information on CVS, or primarily on stable? Can it be provided to customers, or is it more sysadmin oriented?Regards,Greg-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of PaulMahlerSent: Thursday, January 05, 2006 9:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Second edition of my * book has been releasedThe second edition of my Asterisk book "VoIP Telephony with Asterisk" is now in print. It's reorganized and expanded.TKSPaul MahlerPaul Mahler[EMAIL PROTECTED] www.signate.com___--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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--No virus found in this incoming message. Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006___--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 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] How to register a SIP phone on Asterisk behind NAT
Zeeshan a écrit : Hi everybody, One of my client's Asterisk box is behind NAT. They have only one public IP on which they have their router. I can access the Asterisk server using port forwarding (port 22) for SSH. Now this client wants to connect two SIP phones to this Asterisk box from two remote locations. How can this be done. If I forward ports, e.g. 5060-5070 to this Asterisk box, there is no guarantee that the SIP phones will be using the same ports from the remote locations, because ports get changed over the Internet, like for another scenario with my client on public IP, remote SIP ports are 17355, 61949, 61666 etc., though they are configured 5060 on the phones, and 5060 in sip.conf. What is the solution in this scenario of registering SIP on Asterisk behind NAT? Have an Asterisk box on a public IP and use it to glue everything together. Cheers, Jean-Michel. ___ --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] Bridging app
Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. Anyone have any ideas how to do this. I can envision the whole thing except the bridging up the second user. Any assistance, input, or code would be appreciated! Thanks, Wes ___ --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] Easy to Access Telephone Directory AGI
I've written myself a easy to use telephone directory which I use at home and thought it may be of interrest to others. The purpose of this agi script is to provide an online telephone directory that can be easily accessed using the numbers on the phone dial pad. You select entries by spelling out the name of the person you want to contact using the phone dial pad. Now this is normally pretty labourious so the script provides a few shortcuts to make things easier. The best way to illustrate this is by example: Say you want to phone John Smith: - You would start by typing 5, this would find all entries that start with j,k or l. - Next you would type 6 which would narrow down the selection to all tries starting with either j, k or l followed by either m, n or o. - You continue to spell out the name in this fashion (4 = gHi, 6 = mnO, etc) until either a distinct match is found in the direcotry or the number of matches is 9 or less. If a distinct match is found the number associated with the name is returned and can be dialed. If the number of matches is 9 or less you can have an IVR menu containing the matching names built on the fly and you will be prompted to select a name (e.g. Press 1 for John Smith, Press 2 for John Doe etc). Once a name is selected the number associated with the name is returned and can be dialed. Now you might think that this is still pretty laborious but in fact you usually only have to spell out the first few letter of the first name and the last name to get a good match. Other feature include: - Being able to jump to the last name without having to finish spelling out the first name (i.e. Press 0 to skip to the last name) - Multiple numbers can be associated with a name. In this case you will be prompted to select which number you wanted returned for dialing e.g. Press 1 for Home, Press 2 for Business, etc) - Undo last typed entry in case you misstyped something - Wildcard matching (Press 1 to match any letter) - IVR menus built on the fly so you do not need to prerecord anything - IVR menus cached (the more you use it the quicker it gets) - Returns the selected number in the variable DIRNUMBER The code can be found in the Digium Asterisk Users Forum (I was not sure if I should post approx 900 lines of code to this list) http://forums.digium.com/viewtopic.php?t=3727 I can also send it direct if anyone is interrested. ___ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de ___ --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] SIP phones unbeatable echo
Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks in advance. ___ --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] Easy to Access Telephone Directory AGI
Really interesting. Thanks Hannes!! Hannes Vogel wrote: I've written myself a easy to use telephone directory which I use at home and thought it may be of interrest to others. The purpose of this agi script is to provide an online telephone directory that can be easily accessed using the numbers on the phone dial pad. You select entries by spelling out the name of the person you want to contact using the phone dial pad. Now this is normally pretty labourious so the script provides a few shortcuts to make things easier. The best way to illustrate this is by example: Say you want to phone John Smith: - You would start by typing 5, this would find all entries that start with j,k or l. - Next you would type 6 which would narrow down the selection to all tries starting with either j, k or l followed by either m, n or o. - You continue to spell out the name in this fashion (4 = gHi, 6 = mnO, etc) until either a distinct match is found in the direcotry or the number of matches is 9 or less. If a distinct match is found the number associated with the name is returned and can be dialed. If the number of matches is 9 or less you can have an IVR menu containing the matching names built on the fly and you will be prompted to select a name (e.g. Press 1 for John Smith, Press 2 for John Doe etc). Once a name is selected the number associated with the name is returned and can be dialed. Now you might think that this is still pretty laborious but in fact you usually only have to spell out the first few letter of the first name and the last name to get a good match. Other feature include: - Being able to jump to the last name without having to finish spelling out the first name (i.e. Press 0 to skip to the last name) - Multiple numbers can be associated with a name. In this case you will be prompted to select which number you wanted returned for dialing e.g. Press 1 for Home, Press 2 for Business, etc) - Undo last typed entry in case you misstyped something - Wildcard matching (Press 1 to match any letter) - IVR menus built on the fly so you do not need to prerecord anything - IVR menus cached (the more you use it the quicker it gets) - Returns the selected number in the variable DIRNUMBER The code can be found in the Digium Asterisk Users Forum (I was not sure if I should post approx 900 lines of code to this list) http://forums.digium.com/viewtopic.php?t=3727 I can also send it direct if anyone is interrested. ___ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de ___ --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] Dropping incompatible voice frame
I have been getting the following messages on Asterisk for a couple of my client's SNOM phones: 7881 Jan 12 14:52:22 NOTICE[6538] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw ... 8185 Jan 12 14:52:28 NOTICE[6538] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw Messages received for almost 6 secs, generating more than 300 log messages. The next occurrence was an hour later or so, lasting 16 secs, generating more than 800 messages. 8186 Jan 12 15:58:14 NOTICE[6773] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw 8999 Jan 12 15:58:30 NOTICE[6773] channel.c: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to ulaw Sip.conf is setup to use only g729 for these SNOM phones. The SNOMs are also setup to use first g729. G729 licenses are available no problem. There is no error message or anything out of the ordinary preceeding these messages. If any one has any ideas regarding these messages, and what is causing them I would really appreciate a response. Regards to all, Joe ___ --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] Bridging app
On Thu, 2006-01-12 at 13:48 -0600, Schochet, Wes wrote: Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. There might be a simplier way. a channel variable that holds the users response, and a gotoif. You should be able to pass 'g' to dial which according to http://www.voip-info.org/wiki-Asterisk+cmd+Dial g: When the called party hangs up, exit to execute more commands in the current context. So the agent just hangs up and the IVR will continue with the caller into your survey if they so selected, if not it just hangs up. That might be the easiest way to do this. You could even have the agent instructed based on that channel var (depending on your CRM integration) to tell the caller that they will be connected to the survey they opted to do so they dont forget and hangup too. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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] dimensioning: Where is the CPU vs Asterisk load table
Hi, is there any good calculator/table/reference about proper dimensioning? I read the wiki and they basically say xx users run fine in yy hardware http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning. SO far I read that: -Run up to 4 E1s per CPU (which one? an i386 or a dual core? -it is very CPU intensive to do transcoding. Try to minimize it. -you can help the CPU by using DSP-based boards or optimized boards. -Lots of RAM is good too. (like 512MB or 8 GB?). -A Front Side Bus of 800mhz is also good. -disable HT on Intel CPUs. -Use Ram disk to avoid some I/O bottlenecks specially on voicemail (hence, deploy more RAM). -two single core CPUs better than one dual core CPUs??. -And the most important I read was: Keep load under 5 in single CPUs and 10 in dual CPUs (didn't mention dual cores in the article). Im not sure If Im asking properly, basically in this asterisk-heavy-load-learning stage, I want to know how to calculate computer needs based on customer needs. (i've only done 5 to 50 extens, and up to 2 E1) And yes, the need comes from a potential customer that is a Hotel with 450 extensions (rooms) and 125 more ext. (employees) making calls internally, outbound, inboud, voicemail,fax,cell phones, etc. So far, shame on me, I have no idea where to start in terms of equipment. Or I can go out there and buy a 20k machine(s) to run 4or5 E1sIt will run, but I will never learn why. thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] Server Specification
Hi All, I was making plan to set an VoIP Gateway in India. And found some copanies who offered me to host my Asterisk server. I will be appriciated if anyone can suggest me how much simultaneous calls can be handeled with the following server specification? CPU : Dual Intel® Xeon® Processor at 2.8GHz Memory : 512 MB Hard Drive : 2 x 40GB 7.2K RPM Serial ATA Hard Drive Bandwidth : 100GB/MONTH HD Configuration : 2 Hard drives, Motherboard SATA RAID1 : Yes Port : 10/100MBPS SWITCHED VLAN Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] [Announce] Web-MeetMe v2.0.0
[New Features] 1. Added focus and tab-order to all input fields 2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines file includes definitions for the prefered channel and context *** 5. Call history report. Support for this feature requires the php script ./lib/cbEnd.php be running at all times. This also requires a new table in the meetme database if you're upgrading from an earlier release. *** [Location] http://www.fitawi.com/Asterisk [Files] Web-MeetMe_v2.0.0.tgz (required) app_cbmysql.c (required) cbmysql.conf (required) cb-extensions.conf (suggested) README (suggested) [Installation] See the README [Features] 1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blank ii. Identify pin # conflicts (another conference with the same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is set but the User pw is blank d. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference 3. Separate views for Current, Past and Future conferences 4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants 6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added 7. Users can only monitor, update or delete their conferences 8. Verified administrators can monitor, update or delete any conferences. 9. Updated to Asterisk 1.2.0 a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that) Thanks and enjoy, Dan ***Beta testers and anyone who downloaded v2.0.0 before today The only changes from the beta was a cosmetic change to work with non-IE browsers and a couple of installation hints. I only received feedback from one tester, so it appears the package is ready to go. ***Developer help/guidence request*** The PHP script to monitor conference endtime and up date the CDR is fragile. If Asterisk is shut down for more than 30 seconds, the script exits. I'd like to make it more resilent. If any PHP experts can make suggests on how to improve the script it would be appreciated ___ --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: Nested MySQL Commands
Connection pooling doesn't require threading. You can also use a pool of processes which are quite cheap on Linux. Douglas Garstang wrote: Do you have a link to where it says this? The DBI docs that I looked at (perldoc dbi) said that it isn't thread-safe. -Original Message- From: Leo Ann Boon [mailto:[EMAIL PROTECTED] Sent: Thursday, January 12, 2006 12:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Douglas Garstang wrote: I also don't believe perl DBI is thread safe The lastest docs says that DBI does support multithread connection pooling. Otherwise, you are always free to implement your AGI in 'modern' :) programming languages like Java or C# that support threads and pooling. -Original Message- From: Douglas Garstang Sent: Wed 1/11/2006 9:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Re: Nested MySQL Commands Since about 1992... and the Asterisk docs for FastAGI are pretty rotten. But that's ok, I've come to expect that. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Wed 1/11/2006 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands Douglas Garstang wrote: I don't get the whole concept of FastAGI. It's nothing special. Asterisk just opens a connection to a TCP port instead of executing a binary. How long have you been around Unix/Linux systems? Do you have any clue how much less expensive it is to open a TCP socket as compared to forking the Asterisk process, exec()-ing another program, having that program open database/web connections, etc.? ___ --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 ___ --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 ___ --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] Server Specification
well roughly 80 calls on g729 or 120 on g711, figures may differ in realtime, 100 gb bandwidth may not be sufficient, you will have to know the actual throughput too you should check this tool for bandwidth calculation http://www.asteriskguru.com/bandwidth_calculator.php Diyanat From: Abdul Lateef [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Server Specification Date: Thu, 12 Jan 2006 12:09:51 -0800 (PST) MIME-Version: 1.0 Hi All, I was making plan to set an VoIP Gateway in India. And found some copanies who offered me to host my Asterisk server. I will be appriciated if anyone can suggest me how much simultaneous calls can be handeled with the following server specification? CPU : Dual Intel® Xeon® Processor at 2.8GHz Memory : 512 MB Hard Drive : 2 x 40GB 7.2K RPM Serial ATA Hard Drive Bandwidth : 100GB/MONTH HD Configuration : 2 Hard drives, Motherboard SATA RAID1 : Yes Port : 10/100MBPS SWITCHED VLAN Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ --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] interfacing w/ a legacy InterTel PBX
Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for connecting the two systems. I can put another T1/PRI card into the PBX and a Digium TE110P in the Asterisk server. My understanding is that this would give me 23 concurrent conversations between the servers. One question I have is that since I'm using a PRI, will CID information be able to flow through the two systems? Thanks! -Erik Anderson ___ --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] linksys SPA-941
does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term remote provisioning. they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i gave up on those idiots. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] spandsp and page orientation
Shawn, you ever get a fix for this problem? samples are at http://tumtum.no-ip.com/faxes/1128432831.3.tif http://tumtum.no-ip.com/faxes/853107320051004-150908.tif Both of these were faxed from a Brother intellifax 750 through a ring-it single-line simulator into my asterisk box (through an X100P clone) both were normal 8.5X11 pages in portrait style (the map image should be 8.5 wide and 11 long) I can't take the old fax machine offline until I get this resolved. If anyone has any ideas I am open to suggestion. Shawn -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Shawn Porter Sent: Tuesday, October 04, 2005 10:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] spandsp and page orientation I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --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] safe_asterisk not working?
I've been experiencing some crashes in Asterisk in the past few weeks. I haven't been able to find out why as gdb shows it's in a different function every time. But, in the meantime, I've been using safe_asterisk hoping that it would simply restart Asterisk by itself. It doesn't seem to do that. Whenever Asterisk crashed, the list of processes doesn't show asterisk or safe_asterisk running anymore. I do not get an e-mail notification and the core dumped is in the standard format e.g. core.18875 not core.`hostname`-`date -Iseconds`. Does anyone know what I could do to troubleshoot this since I can't really force a crash to see what the script is doing? I have the following variables set in safe_asterisk: #!/bin/shCLIARGS=$* # Grab any args passed to safe_asteriskTTY=9 # TTY (if you want one) for Asterisk to run onCONSOLE=no # Whether or not you want a console NOTIFY=[EMAIL PROTECTED] # Who to notify about crashesDUMPDROP=/tmpI also noticed the line elif [ $EXITSTATUS -gt 128 ];inside the script yet my cores show: Core was generated by `asterisk -vvvg'.Program terminated with signal 11, Segmentation fault.so perhaps that is why the restart is not triggered? ___ --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] linksys SPA-941
Take a look at http://www.sipura.com/support/index.htm.They have an IP Phone administrators Guide that talks about provisioning. It is for the 841 phone but it might give you some hints. Richard Radcliffe Owner Kondor Waffenamt [EMAIL PROTECTED] [EMAIL PROTECTED] 01/12 1:01 pm does anyone get a hold of the SPA-941 Provisioning Guidei tried call Sipuras tech support seems like none ofthem heard of the term remote provisioning. they keptrefering me to their web site which ive check thoroughlyand could not find any documentations on the SPA-941. finallythey gave me a phone number to call which appears to be a faxmachine. thats when i gave up on those idiots.--Edwin Lam [EMAIL PROTECTED]Systems Engineer Office General Inc.Ph: 1 415 439 4988 Fax: 1 415 283 3370http://pgpkeys.mit.edu:11371/pks/lookupop=getsearch=0xD6506D20--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] linksys SPA-941
I don't think they have a specifig Provisioning Guide for each device. They have a general provisionning guide and you can generate an example from the Sipura Profile Compiler for the available options though -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edwin Lam Sent: January 12, 2006 4:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] linksys SPA-941 does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term remote provisioning. they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i gave up on those idiots. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] linksys SPA-941
I have the 941 admin guide in zipped PDF format, I have sent it to a lot of people on the list, if you need it, email me. Cory AndrewsPurchasing Manager++VOIPSupply.comA Division of b2 Technologies454 Sonwil DriveBuffalo, NY 14225direct - 716.250.3402mobile - 716.907.4054email - [EMAIL PROTECTED]AIM - b2Cory - Original Message - From: Radcliffe To: asterisk-users@lists.digium.com Sent: Thursday, January 12, 2006 4:31 PM Subject: Re: [Asterisk-Users] linksys SPA-941 Take a look at http://www.sipura.com/support/index.htm.They have an IP Phone administrators Guide that talks about provisioning. It is for the 841 phone but it might give you some hints. Richard Radcliffe Owner, Kondor Waffenamt [EMAIL PROTECTED] [EMAIL PROTECTED] 01/12 1:01 pm does anyone get a hold of the SPA-941 Provisioning Guide?i tried call Sipura's tech support, seems like none ofthem heard of the term "remote provisioning". they keptrefering me to their web site which i've check thoroughly,and could not find any documentations on the SPA-941. finallythey gave me a phone number to call, which appears to be a faxmachine. that's when i gave up on those idiots.--Edwin Lam [EMAIL PROTECTED]Systems Engineer, Office General, Inc.Ph: +1 415 439 4988 Fax: +1 415 283 3370http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20--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 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
[Asterisk-Users] Zap Channel Error not loading module
Hello I am trying to install the zap channel on asterisk and I get this error == Parsing '/etc/asterisk/skinny.conf': Found Jan 12 15:10:22 WARNING[23463]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [app_controlplayback.so] = (Control Playback Application) == Registered application 'ControlPlayback' [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jan 12 15:10:22 WARNING[23463]: chan_zap.c:933 zt_open: Unable to specify channe l 1: No such device or address Jan 12 15:10:22 ERROR[23463]: chan_zap.c:6525 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jan 12 15:10:22 ERROR[23463]: chan_zap.c:10368 setup_zap: Unable to register cha nnel '1-2' Jan 12 15:10:22 WARNING[23463]: loader.c:345 ast_load_resource: chan_zap.so: loa d_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Jan 12 15:10:22 WARNING[23463]: loader.c:440 load_modules: Loading module chan_z ap.so failed! ___ --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
