RE: [asterisk-users] IVR dictionary dial-plan
From: Steve Kennedy [EMAIL PROTECTED] Date: Mon, 30 Apr 2007 19:33:43 +0100 Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters 2 we know they want bishop if they enter 336 they want demon and 332 they want deacon. There was a similar discussion in the forum, http://forums.digium.com/viewtopic.php?t=14559. Don't seem to have a ready answer. Yuan Liu Could run the dictionary through a script which could generate the dial-plan or do it via some script interactively. Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
I have transitioned to other DID's. I think that company is out of business. Sellvoip is best avoided at all costs. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --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: Anyone having trouble with claling US Domesticon Sellvoip?
I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of people being hauled back to rehab. Anyway, maybe he just makes a habit of running off with people's money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --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] ADSL routers with integrated SIP QoS for other devices
Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which I'll never visit (minimum 3 boxes - ADSL modem, m0n0, WiFi AP). So, does anyone have any recommendations for a wireless ADSL router with integrated QoS for SIP/RTP? I've looked at some of the Draytek units (e.g. Vigor 2700V), but I can't find reference as to whether the integrated QoS applies only to the FXS ports in the router itself, or to all SIP traffic (most of the users will have separate SIP hardphones). These are all to be used in the UK, so the device in question needs to support PPPoA. Any suggestions gratefully appreciated. There are quite a few software stacks for the linksys wrt54g routers/ap's, some of them supportq qos. I hear good things about the wrt routers in general. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 1.4.4 VoiceMail ODBC Storage Help
Bruce McAlister wrote: Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: It looks like the * database user doesn't have permission to delete records from the voicemailmessages table. Make sure it has the propper permissions on that table. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Improving Asterisk's DNS support
Kristian Kielhofner wrote: After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with iffy DNS servers: Maybe that's where you need to start - by fixing the iffy DNS? :-) - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) That's not the responsibility of the application. - dnsmgr doesn't support SIP (yikes!): http://bugs.digium.com/view.php?id=9153 - other randomness (please contribute your own experiences) I haven't been using Asterisk for long, but I have not yet experienced any DNS-related oddities. /Per Jessen, Zürich ___ --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] ADSL routers with integrated SIP QoS for otherdevices
Well on the other side of things there are plenty of adsl equipment running linux and qos capables and customizable firmware. Normally you can get the source of the device with binary drivers of devices like adsl wireless or ethernet switch.. but as long as you stay with the linux version and the tollchain provided you can even compile ztdummy and asterisk to work as a soft pbx. Normally thesse devices are broadcom MIPS based or Texas Instruments AR7 ARM based El Sábado, 28 de Abril de 2007 19:55, Dan Austin escribió: Andrew wrote: On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote: Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD with pf. These are the best solutions, IMO. I was just about to reply with the same recommendation. A SFF chassis with 2 PCI slots could host one S518 and a PSTN interface. These units typically have built-in ethernet and some have built-in wireless. I still have my fingers crossed that Sangoma will offer an ADSL daughercard for the A200. That would make for a perfect combination in a SFF chassis... The latest Linux kernels also have SIP connection tracking/matching, so it should be possible to mark packets and prioritize based on iptables matching. I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not play nice with the wanrouter drivers. (note: there was a recent patch to 2.6.20.4 which apparently has much better SIP matching, and has been tested successfully with Asterisk. I have not tested it yet, and the iptables guys have rejected the patch as their direction for packet matching is shifting significantly in the near future. It can be found at http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18 860.) I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! Good luck and let us know if you find one. The manufacturers of the XDSL chipsets seem to be even worse than the video card companies when it comes to OSS. There's a project on SF called OpenADSL that was working to make common XDSL chipsets work under Linux. The project appears almost dead with a developer post every 6~8 weeks, but that might be a good place to start Looking. If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. It's loosely based off of wondershaper, but works much better, IMO. It does host-based prioritization for VOIP, puts mail just underneath bulk traffic, and P2P beyond that (if you have the p2p connmark stuff set). I can completely saturate DSL links with the S518 with this config without appreciable VOIP degradation. I'm using something similar. The missus can talk to her mother (in rural Japan) over IAX while I am using a IPSEC tunnel to work, and doing heavy downloads. Even without an S518, this script works well with external ADSL/cable modems. You may have to play with the upload rate; some cheap ADSL modems will start blocking your upstream traffic beyond as little as 50% of the upstream rate. ___ --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 -- Francisco J. Pérez Botella___ --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] did we all get spammed by TechnoCo ?
I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. /Per Jessen, Zürich -- http://www.spamchek.com/ - managed email security. ___ --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] headsets for linksys/sipura phones?
I have 4-5 different Nokias, none have a 2.5mm jack. Nothing that even remotely resembles a jack. The older ones did have 2.5 jacks ___ --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 and Billing Issue
Anyone want create a fix for our issue (I will get a price from the client on how much he wants to spend)? Will forcing attended transfers fix this ? - Original Message - From: Jonathan Barratt To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, May 01, 2007 3:37 AM Subject: RE: [asterisk-users] CDR and Billing Issue We have the same problem, and it also showed up when clients made three+ way calls. The CDRs would show them making the same call three times simultaneously as the destination field for the second and third calls was still showing the first number they had dialed. One of our clients caught it and asked us how he could have been calling the same person at the same time as he was already calling them. Quite embarrassing. Not sure if issue has been fixed in subsequent release or not. Best, Jonathan -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Monday, April 30, 2007 2:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR and Billing Issue Hi Guys, I am having an issue that I have been able to replicate and I want to know if anyone else has this. Extension 100 dials an external number. He speaks for 5 minutes and then transfers the call to extension 200. Extension 200 speaks for 1 hour. When we go through the call logs we see the five minute call to the external number from extension 100. We then see a call from extension 100 to extension 200 for 1 hour. The issue we are having is that we are billing the clients (100 and 200 are both the same client as ours) for calls only that hit the PSTN and not internal calls. The issue comes in that if the call is transfer from one extension to another since we see it as a call from one extension to another we assume that it is an internal call. Is there any way to fix asterisk so that it doesn't do this, am I doing some thing wrong or do all calls have to be attended transfers ? (We don't want to tell this to the clients because then they will figure out the loop hole). Thanks a lot. Dovid -- ___ --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] Two Context Residing On The Same Server
I am using the same Asterisk server for 2 different functions. I have users on one side and have a calling platform on one side so I put in a context under general but then only the context for a2billing (calling card platform works) and the other extensions won't work. Below is how I have it set up. [general] context=default ; Default context for incoming calls context=a2billing ;Adding this context for calling card platform___ --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: Anyone having trouble with claling US Domesticon Sellvoip?
Try DIDx.net, I would not say they're best but at least they willing to help you when there is problem and they have a large pool of numbers. -- Original message -- From: Salvatore Giudice [EMAIL PROTECTED] I have transitioned to other DID's. I think that company is out of business. Sellvoip is best avoided at all costs. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --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] CDR and Billing Issue
We had the same problem as well. We ended up blocking all REFER requests on our SIP proxy when the URI was for a PSTN number. A bit inconvenient for customers but preferrable to losing buckets of money. Regards, Grey Man - Original Message From: Jonathan Barratt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 1 May, 2007 1:37:26 AM Subject: RE: [asterisk-users] CDR and Billing Issue We have the same problem, and it also showed up when clients made three+ way calls. The CDRs would show them making the same call three times simultaneously as the destination field for the second and third calls was still showing the first number they had dialed. One of our clients caught it and asked us how he could have been calling the same person at the same time as he was already calling them… Quite embarrassing. Not sure if issue has been fixed in subsequent release or not… Best, Jonathan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Monday, April 30, 2007 2:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR and Billing Issue Hi Guys, I am having an issue that I have been able to replicate and I want to know if anyone else has this. Extension 100 dials an external number. He speaks for 5 minutes and then transfers the call to extension 200. Extension 200 speaks for 1 hour. When we go through the call logs we see the five minute call to the external number from extension 100. We then see a call from extension 100 to extension 200 for 1 hour. The issue we are having is that we are billing the clients (100 and 200 are both the same client as ours) for calls only that hit the PSTN and not internal calls. The issue comes in that if the call is transfer from one extension to another since we see it as a call from one extension to another we assume that it is an internal call. Is there any way to fix asterisk so that it doesn't do this, am I doing some thing wrong or do all calls have to be attended transfers ? (We don't want to tell this to the clients because then they will figure out the loop hole). Thanks a lot. Dovid ___ --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 Send instant messages to your online friends http://au.messenger.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
Re: [asterisk-users] did we all get spammed by TechnoCo ?
Per Jessen wrote: I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a I got it as well. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autoattendant press 1 collides with extension numbers...
So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten = s,1,Background(companyx/companyx-main) exten = s,2,Background(silence/10) exten = s,3,Background(companyx/companyx-main) exten = s,4,Background(silence/10) exten = s,5,Hangup exten = 0,1,Dial(companyx-cust-svce,s,1) exten = 1,1,Goto(companyx-cust-svce,s,1) exten = 2,1,Goto(companyx-shipping,s,1) exten = 3,1,Goto(companyx-accounting,s,1) exten = 4,1,Goto(companyx-sales,s,1) ;exten = 5,1,Goto(companyx-directory,s,1) exten = i,1,Playback(invalid) exten = i,2,Goto(companyx-aa,s,1) [companyx-cust-svce] exten = s,1,Dial(SIP/companyx103,15,tr) exten = s,2,Dial(SIP/companyx100SIP/companyx103,15,tr) exten = s,3,Voicemail([EMAIL PROTECTED]) exten = s,4,Voicemail([EMAIL PROTECTED]) exten = s,5,Hangup -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] chan_local
Hi all, my local channel seems to be not working properly. im doing this: exten= s,1,Dial(Local/[EMAIL PROTECTED],,Tt) some times it rings the phone at extension 123, and sometimes it doesn`t. When it doesnt, it actually displays a msg that it could not find that extension. [May 1 16:54:02] NOTICE[4658]: chan_local.c:563 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel [May 1 16:54:02] WARNING[4658]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Local' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) +its a lie, the user is registered and has extension 12129339038 in context named 'users'++ Can anybody help me. its driving me crazy. i dont know why its doing this and how to solve this problem. im using different extension file for every user and then include every file in a single context named 'users'. i have tried putting all user extensions in a single file but couldent solve the problem. -- Regards Rizwan Hisham Software Engineer ___ --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] Two Connected Servers Sound Quailty
Hi all, I have the same problem using SIP with G729 and it's just on one direction. But ... there is bandwidth management on the FW equipment (sonicwall) and others clients (we are a IP centrex) works find using the same server. A idea ? Thomas De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Matt Gardner Envoyé : samedi, 28. avril 2007 10:06 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Two Connected Servers Sound Quailty Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the best but still isn't reliable) The problems are intermittent sometimes the sound will cut out for 3-4 seconds and other times the sound will just be loosing every other word, and other times it sounds just fine. Also, we have been using Skype over this same Internet connection and have very good sound quality with very few lost words. So here are my questions. First, is it a correct assumption to say that because Skype works well over this connection then I should be able to get asterisk to work over this connect. I am hoping that Skype isn't better then asterisk in this area. If I should be able to get the same sound quality could you point me in the right direction on how to achieve this. (I have tried messing with the jitterbuffer but haven't been able to find very good docs on how to utilize this functionality so about all I have done is set jitterbuffer=yes) 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
[asterisk-users] restrictions on meetme with agi background
I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? How do I bring people into a meetme and play a message to all of them when they are on SIP channels? Jerry ___ --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: restrictions on meetme with agi background
In article [EMAIL PROTECTED], Jerry Geis [EMAIL PROTECTED] wrote: I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? Yes it is. That part of the Meetme architecture hasn't changed at all. How do I bring people into a meetme and play a message to all of them when they are on SIP channels? One way to play an announcement into a conference is to define two extensions - one which Answers and invokes Meetme with the q option, and another which waits for e.g. 1 second and then plays the accouncement. You the create a call, using either a call file or the manager interface. The Channel for the call should be a Local channel that calls the first extension, and the Context/Exten/Priority should specify the second. Another way is to enhance Meetme itself to provide a command to play an announcement file to a specified conference. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
[asterisk-users] My Sip Provider lacks Sip 2.0 183 (Ringing) information
Hi all, my sip provider does'nt send a 183 Message when the opposite party rings. It sends the ringing indication on the audio stream. Is there any chance that the asterisk can analyze this audio stream (meta) information. I saw there is a zaptel configuration entry that sound pretty close to what I need 'callprogress'. Has someone already solved this problem? Knud ___ --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] Autoattendant press 1 collides with extension numbers...
You need to include the context to the extensions (10x) On 5/1/07, J. Oquendo [EMAIL PROTECTED] wrote: So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten = s,1,Background(companyx/companyx-main) exten = s,2,Background(silence/10) exten = s,3,Background(companyx/companyx-main) exten = s,4,Background(silence/10) exten = s,5,Hangup exten = 0,1,Dial(companyx-cust-svce,s,1) exten = 1,1,Goto(companyx-cust-svce,s,1) exten = 2,1,Goto(companyx-shipping,s,1) exten = 3,1,Goto(companyx-accounting,s,1) exten = 4,1,Goto(companyx-sales,s,1) ;exten = 5,1,Goto(companyx-directory,s,1) exten = i,1,Playback(invalid) exten = i,2,Goto(companyx-aa,s,1) [companyx-cust-svce] exten = s,1,Dial(SIP/companyx103,15,tr) exten = s,2,Dial(SIP/companyx100SIP/companyx103,15,tr) exten = s,3,Voicemail([EMAIL PROTECTED]) exten = s,4,Voicemail([EMAIL PROTECTED]) exten = s,5,Hangup -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato ___ --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] Change Codec
Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --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: Voicemail on Different Server (MySQL Replication split thread)
Having master and slave servers in the same switch fabric is the only situation in which I would consider replication. The cases that I described were with machines in separate subnets. Replication simply doesn't work that well when there is significant latency. Did they mention that in your HA class? We setup lab scenarios of replicating databases in different locations, subnets, Master in Dallas, Slave in Chicago, Atlanta. Now that you mentioned that, I do recall a situation whereby high latency between sites could cause replication to be slow, the slaves could fall behind, and with a network connectivity issue data replication could fail. Thanks for the feedback. JR ___ --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] did we all get spammed by TechnoCo ?
Per Jessen wrote: I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. Me too. -- Warm Regards, Lee ___ --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] did we all get spammed by TechnoCo ?
On Tue, May 01, 2007 at 05:36:30AM -0400, Doug Lytle wrote: Per Jessen wrote: I just got spammed by X I got it as well. Same here. However, no point in giving those spammers extra free publicity on the list... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] did we all get spammed by TechnoCo ?
Per Jessen wrote: I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. Ha -- I was just about to post something myself! Yes - I got this too, and immediately suspected a cull of addresses from the mailing list. I'm not impressed. -Stephen- ___ --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] TDM400P and TE405P
Hello All, Is it possible to have both Digium cards installed on one Server (TDM400P and TE405P)? I have one site which requires both connection POT and T1/E1. How can I configure both cards? Thanks, Nitesh ___ --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] Test
where are the out of office replies when they're needed? On 4/30/07, Dovid B [EMAIL PROTECTED] wrote: I love these :) - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 27, 2007 7:54 PM Subject: Re: [asterisk-users] Test Failed On 4/26/07, gc [EMAIL PROTECTED] wrote: ___ --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] did we all get spammed by TechnoCo ?
That stuff is so dangerous. There are too many compliance requirements regarding spam. Doing this kind of stuff opens them up to a lawsuit in more than one state. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, May 01, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] did we all get spammed by TechnoCo ? Per Jessen wrote: I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. Ha -- I was just about to post something myself! Yes - I got this too, and immediately suspected a cull of addresses from the mailing list. I'm not impressed. -Stephen- ___ --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] Autoattendant press 1 collides with extension numbers...
J. Oquendo wrote: So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... Two things: 1 -- your include statement is missing. Asterisk doesn't even know about the extension 101. 2 -- even if you include the other extension, you will have problems, because Asterisk will match in the current context before it checks the included contexts. What this means is that there is no quick fix for your problem. The best thing to do is to make sure that your extensions use a first digit that isn't part of the auto attendant at all. So, for example, your extensions might start with 6 (as in 6XX), since 6 isn't part of the auto attendant menu. Alternatively, you can take the 1 out of the auto attendant. Either way, it will change the caller experience -- but if this is a new setup and it's not working anyway, so what? -Stephen- ___ --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] TDM400P and TE405P
Nitesh Divecha wrote on 5/1/07 10:28 AM: Is it possible to have both Digium cards installed on one Server (TDM400P and TE405P)? I have one site which requires both connection POT and T1/E1. How can I configure both cards? Should work just fine. The Zaptel drivers will pick up both. Just be forewarned, the T1/E1 channels will all get numbered before the POTS channels, no matter what order they're on the bus, so 1-24 will be your T1 and 25-28 the POTS, for example. (I think E1 goes to 32?) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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] Autoattendant press 1 collides with extension numbers...
Stephen Bosch wrote: J. Oquendo wrote: So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... Two things: 1 -- your include statement is missing. Asterisk doesn't even know about the extension 101. 2 -- even if you include the other extension, you will have problems, because Asterisk will match in the current context before it checks the included contexts. What this means is that there is no quick fix for your problem. The best thing to do is to make sure that your extensions use a first digit that isn't part of the auto attendant at all. So, for example, your extensions might start with 6 (as in 6XX), since 6 isn't part of the auto attendant menu. Alternatively, you can take the 1 out of the auto attendant. Either way, it will change the caller experience -- but if this is a new setup and it's not working anyway, so what? -Stephen- ___ --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 Hey Stephen thanks I figured it out... Not worth working on like 2 hours sleep... Actually this was something that was butchered together at the last minute... Changed like 20 times within 2 hours... Not kidding. I had it fine the first time around ... Then re-modified it and forgot the include ;) -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Autoattendant press 1 collides with extension numbers...
Stephen Bosch wrote: Oh and by the way... What i did was... I added a number 5 then sent that to its own context... exten = 5,1,Goto(companyx-directory,s,1) [companyx-directory] exten = s,1,Background(companyx/companyx-directory) exten = 1,1,Dial(SIP/companyx100,15,tr) exten = 2,1,Dial(SIP/companyx101,15,tr) exten = 3,1,Dial(SIP/companyx102,15,tr) exten = 4,1,Dial(SIP/companyx103,15,tr) exten = 5,1,Dial(SIP/companyx104,15,tr) -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] TDM400P and TE405P
Dave Miller wrote: Nitesh Divecha wrote on 5/1/07 10:28 AM: Is it possible to have both Digium cards installed on one Server (TDM400P and TE405P)? I have one site which requires both connection POT and T1/E1. How can I configure both cards? Should work just fine. The Zaptel drivers will pick up both. Just be forewarned, the T1/E1 channels will all get numbered before the POTS channels, no matter what order they're on the bus, so 1-24 will be your T1 and 25-28 the POTS, for example. (I think E1 goes to 32?) Thanks Dave, From the Asterisk CLI when I do zap show status, I get: - hyperion*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3OK 0 0 0 T4XXP (PCI) Card 0 Span 4OK 0 0 0 hyperion*CLI Back of the cards I see four green lights on TDM400P, but no lights on TE405P... And right now I tried calling in on POT lines and Asterisk is not picking up the call... Am I missing something here...? Regards, Nitesh ___ --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 7940 no outgoing audio
Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent but the Cisco can receive voice from the remote phone fine. When we dial Cisco to Cisco it all works fine. I am at a loss to figure this out and any help pointing me in the right direction would be appreciated. We are running an old Asterisk server with version 1.0.10 (yeah we know) and the same mix of hardware and configs works fine. On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco firmware is 08-2-00. Any help appreciated. Regards Simon Alman ___ --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] Change Codec
Put similar allow/disallow statements in the sip or iax entry you create for your outbound ip calls. Be aware that if you use different codecs for phones and your termination provider, all media will have to go through asterisk and you will incur the processing overhead of codec conversion. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arun Kumar Sent: Tuesday, May 01, 2007 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change Codec Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many users can be supported simultaneously?
I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is a Wireless Lan over 802.11b I have checked in wikipedia but I did not find something 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] Email to HP Product Suggestions - Seamless Transparent Fax Gateway
i sent a product suggestion to HP. It was a request to use software that already exists in their JetDirect and Multifunction Fax machines to make them seamlessly interoperate with a fax gateway in a way transparent to the end user. Essentially, giving the sysadmin a choice in fax transport mechanism to route all faxes over analog telephone transport or over the network. Thank you for taking the time to send HP your comments. They have been forwarded to the appropriate people within Hewlett-Packard for their information and review. Should more clarification or information be needed, you may be contacted directly. Sincerely, CEO Customer Relations -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, April 27, 2007 8:26 PM To: External ceo-communication Subject: Feedback to CEO and President Mark Hurd from Robert Townley This is a message to HP's CEO, from a valued customer: Robert Townley. Feedback Area: - Product experience - Ideas and Suggestions Message: SOHOs need seamless transparent fax gateway interoperability. We have been using networkable multifunction fax, copier, scanner, and printer machines since they became available, however the software is only 98% complete. The firmware in these machines needs the following additional options for network faxing and overall paperless office functionality. Essentially another layer between the imaging module and the outgoing fax module so the buyer has a choice of transport mechanism - analog or network. 1.) The systems administrator makes a one time configuration change on the printer itself. The new menu would give the sysadmin the choice of whether faxes go out over traditional analog plain old telephone lines or over the network. In our case, the sysadmin would not even connect wires to the analog telephone line, just the network. 2.) If the sysadmin chooses the new network option, he would would then choose either email/t.37 or IPName/t.38. For the moment, lets just do an internal email setup. He would then enter the email address of the email to fax gateway on the LAN, [EMAIL PROTECTED] 3.) The end user places the paper document he wants to fax into the all-in-one. Enters the fax telephone # he wants to send to, for example 402-555-1234. Presses the Fax button and walks away. Notice the user does NOT enter an IP address or email address, just a fax #. No keyboard required. To the end user, nothing has changed. 4.) The all-in-one sends the TIFF or PDF of the document as an email to [EMAIL PROTECTED] with a subject line of 402-555-1234. 5.) The faxgateway service running on a PC takes the email and faxes it over the offices single analog telephone line to 402-555-1234 using the PC based fax server software. Relevant Links: 1.) $20.00 addon email to fax gateway software for Windows Server 2003. http://www.sandlerco.com/t37fsp.htm 2.) Free and Open Source HylaFax.org email to fax gateway. http://www.hylafax.org/content/Email_to_Fax_Gateway 3.) Free and Open Source Fax Gateway Software http://asterfax.sourceforge.net +1-402-670-4326 This message was sent on: 4/28/2007 at 3:26:25 GMT Robert Townley also added this personal information: Company: iConsultants, PC City: Omaha State: NE Zip Code: 68124 Country:USA Fax:402-391-1233 ___ --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] Delay in Dial()
All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf ___ --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] Calls in ulaw, not gsm as desired
My problem is this We have a location outside of our network which is done over vpn. Everything works except for the voice quality to that location isn't very good. To try to resolve this, I wanted to try to make all calls go over gsm. Right now, when i say show sip channels, they all show ulaw signaling. My setup is pretty basic. I have realtime setup with mysql. In the sip.conf file, I did have disallow=all, allow=ulaw, allow=gsm. I removed those lines from there and in my sip_buddies table, I made sure that the extensions i'm using have disallow=all and allow=gsm. However, even once I reloaded the extensions, its still only using ulaw. Any thoughts? Rob ___ --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] Cisco 7940 no outgoing audio
You should get a packet capture of both cisco-cisco and grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be able to understand the other vendor's devices. BTW, what version of firmware are you running on the cisco phones? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Alman Sent: Tuesday, May 01, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7940 no outgoing audio Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent but the Cisco can receive voice from the remote phone fine. When we dial Cisco to Cisco it all works fine. I am at a loss to figure this out and any help pointing me in the right direction would be appreciated. We are running an old Asterisk server with version 1.0.10 (yeah we know) and the same mix of hardware and configs works fine. On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco firmware is 08-2-00. Any help appreciated. Regards Simon Alman ___ --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] How many users can be supported simultaneously?
The answer is about 42 handsets... Seriously though - you don't mention traffic on the vpn server, you don't mention traffic on the apache server you don't mention anything about transcoding, conference rooms, or if you are using SIP or IAX. You ask an unanswerable question so my answer to any question about the life universe and everything is always...42 Possible you might want to check out here http://www.voip-info.org/wiki/view/Asterisk+dimensioning Or stop being a cheapskate and buy another standalone server for Asterisk :-) Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonopoulos Angelos Sent: Tuesday, 1 May 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How many users can be supported simultaneously? I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is a Wireless Lan over 802.11b I have checked in wikipedia but I did not find something Thanks image001.gif___ --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 1.4.4 VoiceMail ODBC Storage Help
Hi All, I tried to send this email this morning, but I think it has been moderated due to size issue's, so I'll resend it again in 3 parts: PART 1 Hi All, Just an update, after looking a little further into this, it appears that * tries to delete a record that does not exist before inserting it into the table, but the number of times it does that does not match up with the number of warnings displayed on the asterisk console. I have attached the console output, database trace and the actual database definition as text file attachments for readability as well as put it in this email message. Could another pair of eyes please go over this to see if I'm missing anything. Again, if you need any more details from me then please don't hesitate to ask. Asterisk Console Output (verbose 99/debug 99) -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-096da288, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-096da288, voicemail extension=3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/bruce.mcalister-096da288, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-096da288 Playing 'vm-password' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-youhave' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'digits/8' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-and' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'digits/30' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'digits/1' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-Old' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-onefor' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-opts' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-first' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-message' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg.txt': Found -- SIP/bruce.mcalister-096da288 Playing 'vm-unknown-caller' (language 'en') -- SIP/bruce.mcalister-096da288 Playing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-advopts' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-repeat' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-next' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-message' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'digits/2' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0001.txt': Found -- SIP/bruce.mcalister-096da288 Playing 'vm-unknown-caller' (language 'en') -- SIP/bruce.mcalister-096da288 Playing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0001' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-prev' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-advopts' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-repeat' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-next' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-message' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'digits/3' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0002.txt': Found -- SIP/bruce.mcalister-096da288 Playing 'vm-unknown-caller' (language 'en') -- SIP/bruce.mcalister-096da288 Playing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0002' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-prev' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-advopts' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-repeat' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-next' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-delete' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-toforward' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-savemessage' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-helpexit' (language 'en') -- SIP/bruce.mcalister-096da288 Playing 'vm-goodbye' (language 'en') [May 1 07:50:47] WARNING[30791]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] [May 1 07:50:47] WARNING[30791]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] [May 1 07:50:47] WARNING[30791]:
RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help
Hi All, I tried to send this email this morning, but I think it has been moderated due to size issue's, so I'll resend it again in 3 parts: PART 2 Database Table Definition (taken from asterisk readme's) CREATE FUNCTION loin (cstring) RETURNS lo AS 'oidin' LANGUAGE internal IMMUTABLE STRICT; CREATE FUNCTION loout (lo) RETURNS cstring AS 'oidout' LANGUAGE internal IMMUTABLE STRICT; CREATE FUNCTION lorecv (internal) RETURNS lo AS 'oidrecv' LANGUAGE internal IMMUTABLE STRICT; CREATE FUNCTION losend (lo) RETURNS bytea AS 'oidrecv' LANGUAGE internal IMMUTABLE STRICT; CREATE TYPE lo ( INPUT = loin, OUTPUT = loout, RECEIVE = lorecv, SEND = losend, INTERNALLENGTH = 4, PASSEDBYVALUE ); CREATE CAST (lo AS oid) WITHOUT FUNCTION AS IMPLICIT; CREATE CAST (oid AS lo) WITHOUT FUNCTION AS IMPLICIT; CREATE TRUSTED LANGUAGE plpgsql; CREATE FUNCTION vm_lo_cleanup() RETURNS trigger AS $$ declare msgcount INTEGER; begin --raise notice 'Starting lo_cleanup function for large object with oid %',old.recording; --If it is an update action but the BLOB (lo) field was not changed, dont do anything if (TG_OP = 'UPDATE') then if ((old.recording = new.recording) or (old.recording is NULL)) then raise notice 'Not cleaning up the large object table, as recording has not changed'; return new; end if; end if; if (old.recording IS NOT NULL) then SELECT INTO msgcount COUNT(*) AS COUNT FROM voicemailmessages WHERE recording = old.recording; if (msgcount 0) then raise notice 'Not deleting record from the large object table, as object is still referenced'; return new; else perform lo_unlink(old.recording); if found then raise notice 'Cleaning up the large object table'; return new; else raise exception 'Failed to cleanup the large object table'; return old; end if; end if; else raise notice 'No need to cleanup the large object table, no recording on old row'; return new; end if; end$$ LANGUAGE plpgsql; CREATE TABLE public.voicemailmessages ( id BIGSERIAL PRIMARY KEY USING INDEX TABLESPACE bf_service_idx, msgnum SMALLINT NOT NULL DEFAULT 0, dir VARCHAR(80) DEFAULT '', context VARCHAR(80) DEFAULT '', macrocontext VARCHAR(80) DEFAULT '', callerid VARCHAR(40) DEFAULT '', origtime VARCHAR(40) DEFAULT '', duration VARCHAR(20) DEFAULT '', recordingloDEFAULT NULL, mailboxuser VARCHAR(80) DEFAULT '', mailboxcontext VARCHAR(80) DEFAULT '' ) WITHOUT OIDS; CREATE INDEX idx_voicemailmessages_msgnum_dir ON voicemailmessages(msgnum,dir) TABLESPACE bf_service_idx; CREATE TRIGGER trg_vm_cleanup AFTER DELETE OR UPDATE ON voicemailmessages FOR EACH ROW EXECUTE PROCEDURE vm_lo_cleanup(); Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: 01 May 2007 00:06 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help Importance: High Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, Caller VoiceMail Extension = 3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing
RE: [asterisk-users] How many users can be supported simultaneously?
Will you be allowing reinvites? If the server processes media, it will obviously support less simultaneous calls. Also, you may want to rethink the wireless portion. Odds are you will have horrible QoS problems if you run multiple calls or mixed traffic over wireless. BTW, what do you use VPN for? Is this for remote access or for securing VoIP calls? If you are running SIP/G.711, you will need roughly 82.4 kb/s bandwidth. You won't have a bandwidth problem unless you're moving a massive amount of traffic through your VPN or web server. You will likely have a horrible QoS problem. My best guess is that you could push approximately 25 simultaneous calls with no codec conversion, but I wouldn't expect good quality audio. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonopoulos Angelos Sent: Tuesday, May 01, 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How many users can be supported simultaneously? I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is a Wireless Lan over 802.11b I have checked in wikipedia but I did not find something 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 do I do this in Asterisk?
Hi all, I have created a menu from which the caller can select several options such as being transfered to our phones and my mobile phone, meetme, etc. If the caller press an invalid option i have set it to play a message like invalid choice please try again. If the caller make three invalid choices i want the call to be disconnected. what is the best way of doing that? And finally i have set up an extention to which it is possible to record a message but i then want to be able to specify what number the message should be plaied for after recording is finished. Many thanks for all your help, Christian___ --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] Delay in Dial()
Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. Not directly, but yes. Hint: Local channel + Wait. Something like this: Dial(SIP/phoneLocal/[EMAIL PROTECTED]) [delayed] exten = XX,1,Wait(10) exten = XX,2,Dial(SIP/[EMAIL PROTECTED]) --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many users can be supported simultaneously?
Antonopoulos Angelos wrote: I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is a Wireless Lan over 802.11b Exactly 42. ;-) Maybe http://www.voip-info.org/wiki/view/Asterisk+dimensioning http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations http://www.voip-info.org/wiki/view/Asterisk+setup+medium+office+100 is a good point to start reading. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Calls in ulaw, not gsm as desired
It's probably not your codec. Do you have your asterisk box on a Voice VLAN with priority queing configured? If you have mixed traffic on your uplink without VLAN's and priority queuing (or possibly 802.1p), then your QoS will suffer. Changing your codec to GSM will lower bandwidth consumption, but late packets are still late packets. If you can, try to get a measurement of latency to your peering provider before and after setting up QoS. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calls in ulaw, not gsm as desired My problem is this We have a location outside of our network which is done over vpn. Everything works except for the voice quality to that location isn't very good. To try to resolve this, I wanted to try to make all calls go over gsm. Right now, when i say show sip channels, they all show ulaw signaling. My setup is pretty basic. I have realtime setup with mysql. In the sip.conf file, I did have disallow=all, allow=ulaw, allow=gsm. I removed those lines from there and in my sip_buddies table, I made sure that the extensions i'm using have disallow=all and allow=gsm. However, even once I reloaded the extensions, its still only using ulaw. Any thoughts? Rob ___ --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] Delay in Dial()
Hi Steve,
put a timeout in the Dial command, if the call isn't answered it
returns after the timeout has expired
e.g.:
exten = _X.,1,Dial(SIP/${EXTEN}|15)
It waits 15 secs for the call to be answered
Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more
informations
Regards
Edoardo
Steve Finkelstein ha scritto:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a Wait() within a Dial()
would be ideal.
Any suggestions?
- sf
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--
Ing. Edoardo Serra
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Tel: +39 011 678 100
Fax: +39 011 678 275
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[asterisk-users] Channel stuck with call pri flag
Hi, I have a problem where some PRI channels get stuck in a Call mode. If I do a zap show channel XX, it shows as PRI Flags: Call. However there is no calls on that channel. Trying to force a hangup does not work: [EMAIL PROTECTED] Dialer]# asterisk -r -x soft hangup zap/27-1 -- Remote UNIX connection zap/27-1 is not a known channel Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is dundi worth pursuing in this situation?
At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.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] Calls in ulaw, not gsm as desired
However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? -- Andreas Sikkema ___ --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] Delay in Dial()
Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. [extensions] exten = 666,1,Dial(Local/[EMAIL PROTECTED]@Local/[EMAIL PROTECTED]) exten = desk,1,Dial(SIP/deadbeef-a) exten = cell,1,Wait(15) exten = cell,2,Dial(Zap/G1/5551212) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Capture Asterisk traffic
I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --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] TDM400P and TE405P
Hello All, To avoid conflicts I removed TE405P and left the TDM400P and reconfigured the card using genzaptelconf. When I run ztcfg -vv I saw the card and modules are loaded and also I used ztmonitor 1 -v and I saw the gain moving up and down. I did create trunks and outbound routes using FreePBX... Now for some odd reason Asterisk is not picking up the incoming call from PSTN. zapata-channels.conf === ; Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 1 context=default ... zapata.conf #include zapata-channels.conf Can anyone put some light why Asterisk is failing to pickup the call. Regards, Nitesh Dave Miller wrote: Nitesh Divecha wrote on 5/1/07 10:28 AM: Is it possible to have both Digium cards installed on one Server (TDM400P and TE405P)? I have one site which requires both connection POT and T1/E1. How can I configure both cards? Should work just fine. The Zaptel drivers will pick up both. Just be forewarned, the T1/E1 channels will all get numbered before the POTS channels, no matter what order they're on the bus, so 1-24 will be your T1 and 25-28 the POTS, for example. (I think E1 goes to 32?) ___ --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] Test
Test emails and out of office emails make my day. - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 01, 2007 5:37 PM Subject: Re: [asterisk-users] Test where are the out of office replies when they're needed? On 4/30/07, Dovid B [EMAIL PROTECTED] wrote: I love these :) - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 27, 2007 7:54 PM Subject: Re: [asterisk-users] Test Failed On 4/26/07, gc [EMAIL PROTECTED] wrote: ___ --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] Voicemail Creation
You can use real time with an agi. - Original Message - From: mohammad mirzaee To: asterisk-users@lists.digium.com Sent: Sunday, April 29, 2007 12:50 PM Subject: [asterisk-users] Voicemail Creation HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as Extensions but I donot want to add mailboxes in Voicemail.conf Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee -- ___ --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] Calls in ulaw, not gsm as desired
I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Channel stuck with call pri flag
Can you get a layer 3 analyzer trace (e.g., Q.931) and post it here? John Treble Ottawa, Ontario, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: May 1, 2007 12:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Channel stuck with call pri flag Hi, I have a problem where some PRI channels get stuck in a Call mode. If I do a zap show channel XX, it shows as PRI Flags: Call. However there is no calls on that channel. Trying to force a hangup does not work: [EMAIL PROTECTED] Dialer]# asterisk -r -x soft hangup zap/27-1 -- Remote UNIX connection zap/27-1 is not a known channel Any ideas? ___ --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] How do I do this in Asterisk?
Hi Christian,
Increase a variable in the menu loop, or exactly in the t and i
extensions like this :
exten = s,1,Wait(3)
exten = s,n,Answer()
exten = s,n,Set(LoopStep=1)
exten = s,n,Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Wait(1)
exten = s,n(menurestart),Background(your_announce)
exten = s,n,WaitExten(5)
exten = 1,1,GoTo(your_menu_context,1,1)
exten = 2,1,GoTo(your_menu_context,2,1)
exten = 3,1,GoTo(your_menu_context,3,1)
exten = t,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = i,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = h,1,NoOp(the caller has hung up)
I hope that can help and to have not introduced mistakes ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Christian
Envoyé : mardi 1 mai 2007 18:18
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] How do I do this in Asterisk?
Hi all,
I have created a menu from which the caller can select several options such
as being transfered to our phones and my mobile phone, meetme, etc. If the
caller press an invalid option i have set it to play a message like invalid
choice please try again. If the caller make three invalid choices i want the
call to be disconnected. what is the best way of doing that?
And finally i have set up an extention to which it is possible to record a
message but i then want to be able to specify what number the message should
be plaied for after recording is finished. Many thanks for all your help,
Christian
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Re: [asterisk-users] Channel stuck with call pri flag
You mean a PRI debug trace? right now I have some channels that are in this state. There is not much I can do as this is a production system... John Treble wrote: Can you get a layer 3 analyzer trace (e.g., Q.931) and post it here? John Treble Ottawa, Ontario, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: May 1, 2007 12:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Channel stuck with call pri flag Hi, I have a problem where some PRI channels get stuck in a Call mode. If I do a zap show channel XX, it shows as PRI Flags: Call. However there is no calls on that channel. Trying to force a hangup does not work: [EMAIL PROTECTED] Dialer]# asterisk -r -x soft hangup zap/27-1 -- Remote UNIX connection zap/27-1 is not a known channel Any ideas? ___ --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] is dundi worth pursuing in this situation?
I have run into the exact same situation and have the same question. I did it in the dial plan manually due to time contraints but if DUNDi or ENUM or something else is better suited I would love to know. Also the guides and tutorial that I found did not touch on specifics for a situation like this, if anyone knows of one I would be interested in reading it. Thanks, Justin On 5/1/07, Erik Anderson [EMAIL PROTECTED] wrote: At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.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 -- Justin ___ --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] is dundi worth pursuing in this situation?
Erik, Your setup is very similar to one of my own, and I started of manually configuring it, creating IAX connections for each site and then using dial plan to route the call. When I looked at Dundi and finally got it working, I have one IAX connection for all sites and the connections are dynamically created. My dial plan also got simpler, as I add sites I add them to Dundi and the dial plan routes all unmatched extensions to Dundi for lookup. For me dundi has reduced the complexity of my dial plan and I have a pair of servers that query everybody and the that pair listed at my remote sites. I am not using it for least cost routing, yet, but so far it has made things a little easier. You might take a look at the article on txaug.net, under hubguru's articles, it is from JR's Astricon 2006 session. On 5/1/07, Erik Anderson [EMAIL PROTECTED] wrote: At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.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 -- Bruce Reeves Nortex Networks ___ --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] MYSQL application in dial plan
Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? Thanks, __Yehavi: ___ --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] Calls in ulaw, not gsm as desired
Yeah that is fine. You don't need to do any more than that. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] did we all get spammed by TechnoCo ?
Me 3. - Original Message - From: Salvatore Giudice [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, May 01, 2007 5:38 PM Subject: RE: [asterisk-users] did we all get spammed by TechnoCo ? That stuff is so dangerous. There are too many compliance requirements regarding spam. Doing this kind of stuff opens them up to a lawsuit in more than one state. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, May 01, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] did we all get spammed by TechnoCo ? Per Jessen wrote: I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. Ha -- I was just about to post something myself! Yes - I got this too, and immediately suspected a cull of addresses from the mailing list. I'm not impressed. -Stephen- ___ --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] OT: Capture Asterisk traffic
I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)). If a name is used, both the port number and protocol are checked. If a number or ambiguous name is used, only the port number is checked (e.g., dst port 513 will print both tcp/login traffic and udp/who traffic, and port domain will print both tcp/domain and udp/domain traffic). src port port True if the packet has a source port value of port. port port True if either the source or destination port of the packet is port. dst portrange port1-port2 True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value between port1 and port2. port1 and port2 are interpreted in the same fashion as the port parameter for port. src portrange port1-port2 True if the packet has a source port value between port1 and port2. portrange port1-port2 True if either the source or destination port of the packet is between port1 and port2. Any of the above port or port range expressions can be prepended with the keywords, tcp or udp, as in: -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Tuesday, May 01, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Capture Asterisk traffic I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --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] Calls in ulaw, not gsm as desired
Reload will reload your sip.conf file! As well as iax.conf, extensions.conf, queues.conf, voicemail.conf, users.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Capture Asterisk traffic
The RTP traffic is not going to be on port 5060, that is the sip only. Check your rtp.conf file in asterisk for the port range used for RTP traffic. On 5/1/07, CSB [EMAIL PROTECTED] wrote: I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --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 -- Bruce Reeves Nortex Networks ___ --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] TDM400P and TE405P
On Tue, May 01, 2007 at 02:01:33PM -0400, Nitesh Divecha wrote: Hello All, To avoid conflicts I removed TE405P and left the TDM400P and reconfigured the card using genzaptelconf. When I run ztcfg -vv I saw the card and modules are loaded and also I used ztmonitor 1 -v and I saw the gain moving up and down. I did create trunks and outbound routes using FreePBX... Now for some odd reason Asterisk is not picking up the incoming call from PSTN. zapata-channels.conf === ; Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 1 context=default ... zapata.conf #include zapata-channels.conf Can anyone put some light why Asterisk is failing to pickup the call. What do you see in 'zap show channels' in the Asterisk CLI? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] is dundi worth pursuing in this situation?
DUndi or enum only make sense if you plan to move extentions dynamically without having to touch you Asterisk configs or if you want to expose your addressing to the outside world. Personally, I would do it statically so you can avoid delays in processing addressing especially - in the case of enum- if you dns server becomes unavailable. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Tuesday, May 01, 2007 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] is dundi worth pursuing in this situation? At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.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 ___ --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] is dundi worth pursuing in this situation?
DUNDi would be very well suited to this particular application. Publish the extensions that are reachable at each location and when one site dials an extension it gets routed to the one that says i have this. ENUM would probably work just as well for this. I like ENUM with PowerDNS and MYSQL. -Jonathan Justin Hamade wrote: I have run into the exact same situation and have the same question. I did it in the dial plan manually due to time contraints but if DUNDi or ENUM or something else is better suited I would love to know. Also the guides and tutorial that I found did not touch on specifics for a situation like this, if anyone knows of one I would be interested in reading it. Thanks, Justin On 5/1/07, *Erik Anderson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Justin ___ --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] OT: Capture Asterisk traffic
wireshark can further filter out what you don't want, you can also pipe the dump to grep and match only what you want On May 1, 2007, at 11:32 AM, CSB wrote: I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --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] OT: Capture Asterisk traffic
CSB wrote: I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Well, the first thing I notice is that your first tcpdump example is listening on eth0, and the second is listening on eth1. What happens when you do tcpdump -i eth1 -s 0 -w /tmp/tcpdump.1 Do you see the RTP traffic then? -Stephen- ___ --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] My Sip Provider lacks Sip 2.0 183 (Ringing)information
From: Knud Müller [EMAIL PROTECTED] Date: Tue, 01 May 2007 15:19:17 +0200 Hi all, my sip provider does'nt send a 183 Message when the opposite party rings. It sends the ringing indication on the audio stream. Is there any chance that the asterisk can analyze this audio stream (meta) information. I saw there is a zaptel configuration entry that sound pretty close to what I need 'callprogress'. Set progressinband to yes in sip.conf. Yuan Liu Has someone already solved this problem? Knud ___ --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] Delay in Dial()
Eric ManxPower Wieling wrote: Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. [extensions] exten = 666,1,Dial(Local/[EMAIL PROTECTED]@Local/[EMAIL PROTECTED]) exten = desk,1,Dial(SIP/deadbeef-a) exten = cell,1,Wait(15) exten = cell,2,Dial(Zap/G1/5551212) Wouldn't just using the Dial timeout option do the same thing more elegantly? Or do you want the SIP phone to keep ringing? -Stephen- ___ --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] MYSQL application in dial plan
Well, you should be able to leave it open. However, I don't know what would happen if MySQL times out and disconnects the connection because it considers it stale. I don't know if you can check that error and reconnect. Yehavi Bourvine +972-8-9489444 wrote: Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? Thanks, __Yehavi: ___ --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] T1 interface
Would anyone care to recommend a T1 interface method for Asterisk that would function as an (external) alternative to a PCI card like the Digium TE120P? Like some sort of T1-SIP gateway? Also, would anyone with experience using these products care to comment on the practical value of the TE207P vs. the TE205P? smime.p7s Description: S/MIME Cryptographic Signature ___ --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] OT: Capture Asterisk traffic
I remember an app called 'vomit' that could allegedly reconstruct audio files from tcpdump pcap files. Salvatore Giudice wrote: I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)). If a name is used, both the port number and protocol are checked. If a number or ambiguous name is used, only the port number is checked (e.g., dst port 513 will print both tcp/login traffic and udp/who traffic, and port domain will print both tcp/domain and udp/domain traffic). src port port True if the packet has a source port value of port. port port True if either the source or destination port of the packet is port. dst portrange port1-port2 True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value between port1 and port2. port1 and port2 are interpreted in the same fashion as the port parameter for port. src portrange port1-port2 True if the packet has a source port value between port1 and port2. portrange port1-port2 True if either the source or destination port of the packet is between port1 and port2. Any of the above port or port range expressions can be prepended with the keywords, tcp or udp, as in: -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Tuesday, May 01, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Capture Asterisk traffic I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --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] is dundi worth pursuing in this situation?
On 5/1/07, Bruce Reeves [EMAIL PROTECTED] wrote: Your setup is very similar to one of my own, and I started of manually configuring it, creating IAX connections for each site and then using dial plan to route the call. When I looked at Dundi and finally got it working, I have one IAX connection for all sites and the connections are dynamically created. My dial plan also got simpler, as I add sites I add them to Dundi and the dial plan routes all unmatched extensions to Dundi for lookup. For me dundi has reduced the complexity of my dial plan and I have a pair of servers that query everybody and the that pair listed at my remote sites. I am not using it for least cost routing, yet, but so far it has made things a little easier. You might take a look at the article on txaug.net, under hubguru's articles, it is from JR's Astricon 2006 session. Thanks for the info, Bruce. It sounds like it would be at least worth giving Dundi a try. I've never touched it before now, but I can't imagine a configuration like mine would be too complex. We'll see how this goes... -erik ___ --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] Improving Asterisk's DNS support
I haven't been using Asterisk for long, but I have not yet experienced any DNS-related oddities. Then keep using it, and you will. ___ --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] Calls in ulaw, not gsm as desired
That's what I did though. So my sip.conf file no longer has any allows in it. Instead, it should be relying on the realtime settings for that. However, even though I told it to only use 5053, it still is using ulaw. Rob Salvatore Giudice wrote: Yeah that is fine. You don't need to do any more than that. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Schall *Sent:* Tuesday, May 01, 2007 2:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] chan_sip seems to be hanging
I posted about this problem last week and thought it was a combination of SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the SIP channel that's hanging. When this happens a call can still come in and hit the IVR, but no one can connect to the server from a SIP client. I tried reloading chan_sip.so today when this occurred, and I tried unloading chan_sip.so but was told the channel was in use. How can I clear SIP connections? With ZAP channels I can use ZAP DESTROY CHANNEL, but I don't see the equivalent for SIP. Any suggestions for tracking down what's causing SIP to hang? My only option as it stands is to shutdown asterisk restart it, I included a piece of the log last week and am willing to do so again if needed. If I can see which SIP channels the server thinks are open when the channel hangs I'm hoping this will allow me to find if it's a common phone or perhaps some dialplan logic gone bad. Thanks, Ken ___ --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] Delay in Dial()
Stephen Bosch wrote: Eric ManxPower Wieling wrote: Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. [extensions] exten = 666,1,Dial(Local/[EMAIL PROTECTED]@Local/[EMAIL PROTECTED]) exten = desk,1,Dial(SIP/deadbeef-a) exten = cell,1,Wait(15) exten = cell,2,Dial(Zap/G1/5551212) Wouldn't just using the Dial timeout option do the same thing more elegantly? Or do you want the SIP phone to keep ringing? No. Dial(SIP/deskSIP/cell) would find BOTH phones at the same time. The original poster wants the desk phone to ring, then after X seconds, KEEP ringing the desk phone, but also ring the cell phone. Using the Dial timeout would STOP ringing the desk phone then, start ringing the desk phone again and also ring the cell phone. I don't know about you but it would seem pretty unprofessional to me if my deskphone rang, I went to pick it up, got a dialtone because I did not get to it in time, then before I hungup the deskphone Asterisk rang both the desk and cell phone. Since the deskphone is offhook the call could go immediately to voicemail and then there would be no call when you rushed over to pick up the cell phone. ___ --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] Stanaphone business ok?
I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble thanks Todd ___ --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] Display Caller ID of called party
Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in case that matters. Any ideas? ___ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] SRTP implementation
Olle E Johansson wrote: 23 apr 2007 kl. 19.55 skrev Russell Bryant: John Todd wrote: To morph this into a -dev thread: if this patch were to become (again) useful and error-free, is there any objection or usefulness in adding it to TRUNK? Personally, I think there is, if there is a method by which SRTP can be activated or de-activated from within the dialplan based on prior shared secrets. However, I have heard others disagree and object that without signalling-based secure key exchange, SRTP is not worth the effort. Opinions? I agree with you. I think that is a reasonable approach. I can't speak for the quality of the patch itself as I have not reviewed it. But, if it works, I would guess that it would not be too bad to get it into trunk. Kevin and I earlier decided that we wanted to delay this until we had a complete security solution, with signalling based secure key exchange ;-) /O I have uploaded a new patch. This patch and also the previous supports MIKEY as well as sdescriptions. The MIKEY key management scheme uses transport encryption for transporting the keys securely over unsecured transports such as unencrypted SDP. There are several MIKEY flavors: Pre shared, DH-SIGN, RSA, RSA-R and DH-HMAC. The patch currently uses DH-HMAC for outgoing connections, using secret from sip.conf as the shared secret. http://www.voip-info.org/wiki/view/Asterisk+SRTP updated test srtp server (asterisk SVN-trunk-r61760 + latest SRTP patch) voice2.fpf.slu.cz test sip accounts 700:700 701:701 702:702 extensions.conf exten = 600,1,Set(_SIPSRTP=optional) exten = 600,n,Set(_SIPSRTP_CRYPTO=enable) exten = 600,n,Playback(demo-echotest) ; Let them know what's going on exten = 600,n,Echo ; Do the echo test exten = 600,n,Playback(demo-echodone) ; Let them know it's over exten = 600,n,hangup exten = 610,1,Set(_SIPSRTP=require) exten = 610,n,Set(_SIPSRTP_MIKEY=enable) exten = 610,n,Playback(demo-echotest) ; Let them know what's going on exten = 610,n,Echo ; Do the echo test exten = 610,n,Playback(demo-echodone) ; Let them know it's over exten = 610,n,hangup p.s. sorry for cross post --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --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] Delay in Dial()
It's amazing how simple some answer are.
Thank you kindly for your responses Edoardo and Luki. :-)
- sf
Edoardo Serra wrote:
Hi Steve,
put a timeout in the Dial command, if the call isn't answered it
returns after the timeout has expired
e.g.:
exten = _X.,1,Dial(SIP/${EXTEN}|15)
It waits 15 secs for the call to be answered
Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more
informations
Regards
Edoardo
Steve Finkelstein ha scritto:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a Wait() within a Dial()
would be ideal.
Any suggestions?
- sf
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Re: [asterisk-users] MYSQL application in dial plan
Yehavi Bourvine +972-8-9489444 wrote: Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? or, you can use func_odbc that comes with * 1.4. Now you don't have to connect to your db every time you use it, I think. Thanks, __Yehavi: ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] T1 interface
You could get yourself a cisco universal gateway or a Audiocodes Mediant 1000 Single Span T1 SIP Gateway. With regard to the cards: In my experience, you want an echo cancellation card if you are connected to a carrier without echo cancellers. Typically, LEC circuits do not have echo cancellers and long distance carriers do. I personally do not buy Digium hardware anymore. I've had such an abysmal experience with Digium's hardware quality and overall support in th past that I now only use Sangoma equipment. I have never had a problem with Sangoma's equipment. Their service is exemplary and they have even offered me free professional services in the past to optimize my gateway setup. I wouldn't spit on Digium hardware if it was on fire. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: Tuesday, May 01, 2007 3:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] T1 interface Would anyone care to recommend a T1 interface method for Asterisk that would function as an (external) alternative to a PCI card like the Digium TE120P? Like some sort of T1-SIP gateway? Also, would anyone with experience using these products care to comment on the practical value of the TE207P vs. the TE205P? ___ --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] OT: Capture Asterisk traffic
Ethereal will let you export an rtp stream as a .au file. That's one of the very minor items we cover in our conference series and our VoIP 100 course. There is a lot more fun to be had when you get into RTP sequence number prediction and RTP stream I injection. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Garstang Sent: Tuesday, May 01, 2007 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Capture Asterisk traffic I remember an app called 'vomit' that could allegedly reconstruct audio files from tcpdump pcap files. Salvatore Giudice wrote: I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)). If a name is used, both the port number and protocol are checked. If a number or ambiguous name is used, only the port number is checked (e.g., dst port 513 will print both tcp/login traffic and udp/who traffic, and port domain will print both tcp/domain and udp/domain traffic). src port port True if the packet has a source port value of port. port port True if either the source or destination port of the packet is port. dst portrange port1-port2 True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value between port1 and port2. port1 and port2 are interpreted in the same fashion as the port parameter for port. src portrange port1-port2 True if the packet has a source port value between port1 and port2. portrange port1-port2 True if either the source or destination port of the packet is between port1 and port2. Any of the above port or port range expressions can be prepended with the keywords, tcp or udp, as in: -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Tuesday, May 01, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Capture Asterisk traffic I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --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] is dundi worth pursuing in this situation?
Salvatore Giudice wrote: DUndi or enum only make sense if you plan to move extentions dynamically without having to touch you Asterisk configs or if you want to expose your addressing to the outside world. Personally, I would do it statically so you can avoid delays in processing addressing especially - in the case of enum- if you dns server becomes unavailable. I've been using both enum and dundi. Dundi has some means of setting one server to primary, so that server will most likely have the number you are looking for in it's cache. With enum, you'll want to run a dns recursor on each * host or on a host very close to it networkwise. Both will do equally well in your case. I like the easy of using dundi. There are some very good dialplans floating around for doing enum lookups, I have a macro written in ael2 that you can have if you like. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Calls in ulaw, not gsm as desired
How did you set it to 5053? Can you post your sip.conf? You should remove the passwords and ip addresses, etc. Usually, it's just an allow and a disallow statement inserted into each inbound and outbound channel definition. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired That's what I did though. So my sip.conf file no longer has any allows in it. Instead, it should be relying on the realtime settings for that. However, even though I told it to only use 5053, it still is using ulaw. Rob Salvatore Giudice wrote: Yeah that is fine. You don't need to do any more than that. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Delay in Dial()
Eric ManxPower Wieling wrote: Stephen Bosch wrote: Eric ManxPower Wieling wrote: Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. [extensions] exten = 666,1,Dial(Local/[EMAIL PROTECTED]@Local/[EMAIL PROTECTED]) exten = desk,1,Dial(SIP/deadbeef-a) exten = cell,1,Wait(15) exten = cell,2,Dial(Zap/G1/5551212) Wouldn't just using the Dial timeout option do the same thing more elegantly? Or do you want the SIP phone to keep ringing? No. Dial(SIP/deskSIP/cell) would find BOTH phones at the same time. The original poster wants the desk phone to ring, then after X seconds, KEEP ringing the desk phone, but also ring the cell phone. Using the Dial timeout would STOP ringing the desk phone then, start ringing the desk phone again and also ring the cell phone. and how long in seconds would you think it takes * to step from the first dial to the second? Is this a real risk? I don't know about you but it would seem pretty unprofessional to me if my deskphone rang, I went to pick it up, got a dialtone because I did not get to it in time, then before I hungup the deskphone Asterisk rang both the desk and cell phone. Since the deskphone is offhook the call could go immediately to voicemail and then there would be no call when you rushed over to pick up the cell phone. ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Stanaphone business ok?
Write them and ask. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd H Sent: Tuesday, May 01, 2007 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Stanaphone business ok? I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble thanks Todd ___ --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] Stanaphone business ok?
Hmmm that's not good, I've been very happy using them as a backup line to my packet 8 services. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Todd H Sent: Tuesday, 1 May 2007 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Stanaphone business ok? I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble thanks Todd ___ --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] Display Caller ID of called party
Kevin, It seems to me that what you are really talking about is manipulating the display features of the phone. Caller ID is unlikely to have this effect as the phone does not consider the From: URI in the SIP header unless the call is of an incoming nature. The solution to this is bound to be proprietary to the phone in some way or another--if there is one. I just wanted to point out that the mechanism for its delivery would almost certainly not be caller ID. Of course, you COULD always set your dial plan in such a way that it never actually completes the outbound call leg, but instead hangs up, and then dials it, and rings you back (with the caller ID of the intended incoming leg). -- Alex -- Alex Balashov [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] Display Caller ID of called party
On 5/1/07, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote: Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in case that matters. On a Grandstream GXP-2000, this happens when the dialed number is in the XML phonebook that the phone sucks down from my provisioning server. It might work the same on the Polycom units. Of course, you need to have some process in place to keep the phonebook file up to date. To do it in a generic way where the name is looked up by * and sent back to your phone for display as part of the 100 Trying or 180 Ringing responses, is an entirely different matter. I suspect that the end-user experience would vary wildly based on the equipment each user was using. If this is possible, I'm sure people more knowledgeable than me will chirp in. The phonebook route might be the quickest bang for your buck though. If I recall from testing Polycom phones, you can have a central phonebook shared by all phones and a per-phone phonebook that is uploaded by the phone to your TFTP server so that even when re-provisioning from factory reset, nothing is lost. I didn't get far enough in the evaluation to set up a provisionin server of my own. The evaluation died in committee when an exec reported that she didn't like the small buttons on the IP430. -- j. ___ --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 kernel module load order
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mitch Jackson wrote: Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which happens more often, nothing will initialize properly. When this happens, I have to unload all the zaptel modules, and re-load them over and over again, until the hardware comes up in the correct order. The order it is loaded is in no way related to what order I load the modules on the command line. This problems makes it unlikely that asterisk will start properly if the system is rebooted. Is there something I can do to ensure the modules get loaded in the correct order? If you use udev (and subsequently modprobe), you can override the install command for the TDM card to load the T1 card first: Create a file in the /etc/modprobe.d directory, and put the following line in it: install wctdm4xxp modprobe wcte21xp modprobe --ignore-install wctdm4xxp $CMDLINE_OPTS /sbin/ztcfg (Make sure it's all one line; your mail reader might break it.) This method works for me, with a TE110P and a TDM2400P in the same box. However, I am using Debian, and I'm not sure if modprobe and udev work the same way in FC6. TTYL. - -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGN7bcKeSNHCYiCKARAoefAKDAh/V2W3cwd/ASfHH5JsMOdj3wOgCgnexb yMh5TUnMZzHdM572J67oxmU= =pHCB -END PGP SIGNATURE- ___ --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
