Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)
On 2 Apr 2006, at 04:27, Rich Adamson wrote: Kevin P. Fleming wrote: Rich Adamson wrote: Is this worthy of opening a bug assuming the above comment is still valid? Would the individual(s) maintaining res_snmp want to log into either of these internet accessible boxes to identify the root cause? The module loader in trunk is undergoing changes that will eliminate this problem very soon. No problem. Thanks. For what it is worth, I put some (badly formatted) notes on the wiki describing how I got res_snmp working. http://www.voip-info.org/wiki/view/Asterisk +monitoringview_comment_id=10174 Since then we have added some asterisk specific example beans to westhawk's java SNMP stack (http://snmp.westhawk.co.uk) and we are working on a little monitoring GUI which we will release once we have it looking sensible :-) ! res_snmp doesn't tell you much that you couldn't find via the manger interface, but it is READ_ONLY and if you use SNMPv3 it is reasonably secure. Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 nat problems.
I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksyswrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone directly into the cable modem and now the phone can connect fine and works. I pasted below the sip debug output, anybody know what's going on or have experience with this? sip.conf [general]context=defaultbindport=5060bindaddr=0.0.0.0 srvlookup=yes [1002]username=1002secret=type=friendhost=dynamicallow=allcontext=defaultnat=yes -- SIP DEBUG - -- SIP read from 68.5.xxx.xxx:51065:INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5fFrom: "1002" sip:[EMAIL PROTECTED];tag=00115cd9d0370002128504be-71e03bcbTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Max-Forwards: 70CSeq: 101 INVITEUser-Agent: Cisco-CP7960G/8.0Contact: sip:[EMAIL PROTECTED]:5060;transport=udpExpires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATESupported: replaces,join,norefersubContent-Length: 274Content-Type: application/sdpContent-Disposition: session;handling=optional v=0o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102s=SIP Callt=0 0m=audio 25584 RTP/AVP 0 8 18 101c=IN IP4 192.168.1.102a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/0a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecv --- (16 headers 13 lines)---Using INVITE request as basis request - [EMAIL PROTECTED]Sending to 192.168.1.102 : 5060 (non-NAT)Reliably Transmitting (NAT) to 68.5.xxx.xxx:51065:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxxFrom: "1002" sip:[EMAIL PROTECTED];tag=00115cd9d0370002128504be-71e03bcbTo: sip:[EMAIL PROTECTED];tag=as0ed772bfCall-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msFound user '1002'localhost*CLI-- SIP read from 68.5.xxx.xxx:51065:INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5fFrom: "1002" sip:[EMAIL PROTECTED];tag=00115cd9d0370002128504be-71e03bcbTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Max-Forwards: 70CSeq: 101 INVITEUser-Agent: Cisco-CP7960G/8.0Contact: sip:[EMAIL PROTECTED]:5060;transport=udpExpires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATESupported: replaces,join,norefersubContent-Length: 274Content-Type: application/sdpContent-Disposition: session;handling=optional v=0o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102s=SIP Callt=0 0m=audio 25584 RTP/AVP 0 8 18 101c=IN IP4 192.168.1.102a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/0a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecv --- (16 headers 13 lines)---Ignoring this INVITE requestRetransmitting #1 (NAT) to 68.5.xxx.xxx:51065:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxxFrom: "1002" sip:[EMAIL PROTECTED];tag=00115cd9d0370002128504be-71e03bcbTo: sip:[EMAIL PROTECTED];tag=as0ed772bfCall-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm="asterisk", nonce="74f7630a"Content-Length: 0 ---localhost*CLI-- SIP read from 68.5.xxx.xxx:51065:INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5fFrom: "1002" sip:[EMAIL PROTECTED];tag=00115cd9d0370002128504be-71e03bcbTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Max-Forwards: 70CSeq: 101 INVITEUser-Agent: Cisco-CP7960G/8.0Contact: sip:[EMAIL PROTECTED]:5060;transport=udpExpires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATESupported: replaces,join,norefersubContent-Length: 274Content-Type: application/sdpContent-Disposition: session;handling=optional v=0o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102s=SIP Callt=0 0m=audio 25584 RTP/AVP 0 8 18 101c=IN IP4 192.168.1.102a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/0a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecv --- (16 headers 13 lines)---Ignoring this INVITE requestRetransmitting #2 (NAT) to 68.5.xxx.xxx:51065:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxxFrom: "1002" sip:[EMAIL
Re: [Asterisk-Users] H323 on way voice
This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI h.323 debugH323 debug enabledmygw== New H.323 Connection created. -- Setting up Call -- oCall token: [ip$192.168.1.219:1057/8226] -- oCalling party name: [myPersonal] -- oCalling party number: [] -- oCalled party name: [ip$192.168.1.214:1720] -- oCalled party number: []mygw--Received SETUP messageAllowed Codecs: mygw Table: G.729A 1 G.729 2 G.723.1 3 G.711-uLaw-64k 4 G.711-ALaw-64k 5 UserInput/hookflash 6 UserInput/RFC2833 7 Set:aco*CLI 0:aco*CLI 0:o*CLI G.729A 1 G.729 2 G.723.1 3 G.711-uLaw-64k 4 G.711-ALaw-64k 5 1:o*CLI UserInput/hookflash 6 2:o*CLI UserInput/RFC2833 7mygw*CLImygw=-= In OnAnswerCall for call 8226mygw*CLI - Progress Indicator: 0mygw*CLI - Inserting PI of 0 into ALERTING message == Starting H323/ip$192.168.1.219:1057/8226 at default,ip$192.168.1.214:1720,1 failed so falling back to exten 's' -- Executing Playback(H323/ip$192.168.1.219:1057/8226, demo-echotest) in new stack mygwAnswering call ip$192.168.1.219:1057/8226 -- Playing 'demo-echotest' (language 'en')mygw-- Started logical channel: sending G.711-uLaw-64kmygw*CLI -- channelsOpen = 1mygw=-= In OnConnectionEstablished for call 8226 mygw*CLI -- Connection Established with myPersonal [192.168.1.219]mygwMyH323_ExternalRTPChannel::OnReceivedAckPDUmygw*CLI -- remoteIpAddress: 192.168.1.219 -- remotePort: 49600 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.219 -- remotePort: 49600 -- ExternalIpAddress: 192.168.1.214 -- ExternalPort: 17950mygw*CLI . -- Executing Echo(H323/ip$192.168.1.219:1057/8226, ) in new stack ___ --Bandwidth 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] FreePBX on Debian
On 4/2/06, Christian Gröger [EMAIL PROTECTED] wrote: need that Zaptel stuff? It always prompted errors so i am now using mISDN -without errors, is there a module for freePBX for mISDN? to use mISDN with freepbx, you can Add custom trunk in the Trunks menu. Anyway, is there a good manual for installing FreePBX on debian? Something with typical debian-errors and stuff? That standard manual is so focussed on Suse :( Try the readme or INSTALL file, in the archive. It basically explains what you need to do. Or join #freepbx on irc. cheers ___ --Bandwidth 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] H323 on way voice
I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. Michael Mansos(or something like that) and other guys have been done a good job. Isamar ___ --Bandwidth 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] Line Pick up Problem
Could anyone shed some light on this problem: Running Asterisk @ Home 1.7 When call comes in (Zap Clone 100 Card), the extensions ring but when lifting the receiver on an IP phone (BudgeTone 100), I just hear a bit of crackle sound in time with the ringing of the other extensions. The external caller still has the ringing sound, so asterisk has not picked up the line. If I set Immediate Answer to Yes, then Asterisk picks up the line, the extensions ring and after lifting the receiver of an IP phone the call is completed (i.e. can talk to the caller). It doesn't make sense that asterisk knows how to answer the line in the Immediate Answer Mode but if it does it without Immediate Answer and the ringing extension's receiver is lifted it fails. I've rechecked all configurations (both phone and Asterisk) as per setup guides and kept everything as standard as possible. I've no idea why I can't have the line answered after the extension's receiver is lifted. Any ideas on how to fix this? Thanks in advance, Richard attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom overlap dialing?
Is there any way to get a polycom 601 to do overlap dialing? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom overlap dialing?
Is there any way to get a polycom 601 to do overlap dialing? I can't find anything on the subject, which confirms my initial hunch: I really doubt it. You could probably work something up in asterisk, though. - Noah ___ --Bandwidth 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: Polycom IP501 and Speed Dials
Hi Again Avi - Sadly, that doesn't work -- the Polycoms store their directories locally as well and re-upload them on reboot. Another idea: Can you create the mac address-directory.xml files as symlinks to the central file? Maybe if the phone sees a directory file already there it will not overwrite it with the settings in memory. - Noah ___ --Bandwidth 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] Building Asterisk embedded device
Hi Sam - Thanks for your link. how to build asterisk into this hardware? As mentioned earlier, have a look at astlinux: http://www.astlinux.org/ There are pre-built versions for soekris/wrap, and general x86 computers. Kristian (the astlinux developer) made this run on a gumstix, too, but I don't think there's a pre-built version of that. You could ask Kristian directly how to do this. - Noah ___ --Bandwidth 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 box with unreliable ping/latency
Hi Bjorn - Everything you mentioned seems to point to the problem being a hardware issue, or more specifically the way that FC and CentOS are using your hardware. Why not use different hardware and/or OS? Maybe FC and CentOS just use faulty driver for your NIC? - Noah ___ --Bandwidth 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 7960 nat problems.
Shaun wrote: I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone directly into the cable modem and now the phone can connect fine and works. I pasted below the sip debug output, anybody know what's going on or have experience with this? You need to turn on the NAT support for the 7960 on the phone itself. Settings -- SIP Configuration --NAT Enabled (Y/N) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec problems
Or, could he be seeing an outstanding issue with counting (eg, start with 0 or 1)? Seems like that might be the case. I've got about ten g729 licenses and never see any warning messages, but then again this is a small system and I don't think I could consume all of them if I tried. Alyed Tzompa wrote: I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got. If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk. Also, if you have enough time try using the g729 with another soft / IP phone and see if you get the same result. Alyed --- I am not. I have one license and use i channel. It seems to detect the fact there are no more channels left and keeps warning me about it in case I want to use more. It is fine, but the warning is constant. All you see on Asterisk console is running warning message. Rudolf On 4/2/06, Kevin P. Fleming wrote: RumaTech wrote: And it keeps running like that. Call usually come through OK. If i try to use show g729 command, it shows that all codecs are in use. Well, this is fine, I am using one, but I do not want to see those warnings. Once is quite enough. Those continuos warnings make it impossible to se any other asterisk output. How does one turns them off? You can't make them stop except by not trying to use more channels than you have licenses for. ___ --Bandwidth 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] Re: How is Teliax ?
end-to-end path. Each step through the tracert process does nothing more then issue an icmp echo request, measuring the response time and displaying it. maybe on windows it does icmp echo but no unix does this (at least not by default). i recommend you study what unix traceroute actually does. :) I'm very heavy (professionally) into protocol analysis, and yes unix does rely on icmp to perform the traceroute. (icmp pkt type 11, code 0) If you're a non-believer, put an access list on all icmp traffic and see if your traceroute continues to function. :) ___ --Bandwidth 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] Install problem with res_snmp.so from current trunk (bug?)
Rich Adamson wrote: Is this worthy of opening a bug assuming the above comment is still valid? Would the individual(s) maintaining res_snmp want to log into either of these internet accessible boxes to identify the root cause? The module loader in trunk is undergoing changes that will eliminate this problem very soon. No problem. Thanks. For what it is worth, I put some (badly formatted) notes on the wiki describing how I got res_snmp working. http://www.voip-info.org/wiki/view/Asterisk+monitoringview_comment_id=10174 Since then we have added some asterisk specific example beans to westhawk's java SNMP stack (http://snmp.westhawk.co.uk) and we are working on a little monitoring GUI which we will release once we have it looking sensible :-) ! res_snmp doesn't tell you much that you couldn't find via the manger interface, but it is READ_ONLY and if you use SNMPv3 it is reasonably secure. I posted the original message because two different versions of the libnetsnmp existed on the two systems, and the makefile indicated the author was having issues that were suspected to be version dependent. So I was offering up a system that had the problem (both fc3) in order to assist in identifying the root cause. I really don't have an interest of actually implementing snmp at this time; just trying to help identify issues where I can as a non-programmer. ___ --Bandwidth 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 7960 nat problems.
Shaun wrote: I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone directly into the cable modem and now the phone can connect fine and works. I pasted below the sip debug output, anybody know what's going on or have experience with this? I have several of these working just fine. You really do need nat=yes in sip.conf, and, change the 7960 config to say nat=yes in the 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
Re: [Asterisk-Users] H323 on way voice
Hello, As I know from my experience with Chan323 and OH323. I setup Asterisk on Redhat 9.0 i386 and it is working without any problem with Chan323, OH323 libraries required. I never tried OOH323 come (0.4) with Asterisk. If possible I would like to know how to use newest version of OOH323 (0.8.1) with Asterisk? boldsoft*CLI show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686 running Linux Interesting thing is even sometimes X-Lite doesn't work properly sometimes if both end behind NAT. But If I to use X-Pro one-end and other-end X-Lite, this case working normal. But Audiocodes, Addpac, Davolink and other gateways with G729, G723 working without any problem with Chan323 and OH323. I did upgrade Asterisk from existing version to latest 1.2.6 and installed Chanh323 and OH323 0.7.3 with neccessary libraries. Both work one-way voice only, when I to use X-Lite and X-Pro. I don't know how to get work Chan323 and OH323 with Asterisk 1.2.6. Regards, Balgaa - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 02, 2006 8:08 PM Subject: Re: [Asterisk-Users] H323 on way voice I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. Michael Mansos(or something like that) and other guys have been done a good job. Isamar ___ --Bandwidth 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] vmail access problem
Evar,I tried to duplicate this on my system but wasn't able to. Do you have the same password set for both voicemail boxes? Have you already logged into both of them individually (maybe it's a cookie thing?) KyleOn 4/1/06, Ever Zalazar [EMAIL PROTECTED] wrote: Hi everybody..I have the follow problem with my vmail access: http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action="" For example this is the address to access the voice mail of one customer. If that customer change the number for : http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action="" He will access that user account and see the messages.HOw can I protect this? Thanks Ever ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Steve Gladden wrote: What version of asterisk? (been lots of changes happening to the sip code over the last year) SVN-branch-1.2-r9156 Have you looked at the sample configs in /usr/src/asterisk/configs? Yes I have and my own configs are pretty much copies of them. They do not detail, do or explain the simple concept that I am trying to accomplish. If they do I don't see it. #1 I have more than one incoming SIP account #2 I would like to have them come into the context of my choice when a call comes in. HOW do I do this? currently I have 3 register lines there is no way to specify in a register line some way of making the call start in any other context other than what is specified in the [general] section of sip.conf It seems that somehow maybe if there is a peer tat is somehow matched to the register line (how???) it may work. There may be some crazy way to do this within a peer if so this is the information I am looking for... The examples and descriptions are not at all clear to me I have 3 accounts with the same provider How do I get incoming calls to come into three different contexts that I will create is the question. From the example file I see: Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register = user[:secret[:[EMAIL PROTECTED]:port][/extension] ; ; If no extension is given, the 's' extension is used. The extension needs to ; be defined in extensions.conf to be able to accept calls from this SIP proxy I actually need to do 3 of these. ;register = 2345:[EMAIL PROTECTED]/1234 ; ;Register 2345 at sip provider 'sip_proxy'. Calls from this provider ;connect to local extension 1234 in extensions.conf, default context, ;unless you configure a [sip_proxy] section below, and configure a ;context. Ok I have 3 accounts from the same provider one [sip_proxy] section just puts me in the same problem boat I'm already in using a register line the calls some into the context specified in [general] section of sip.conf I need to somehow differentiate the three SIP 'lines' and give them different contexts to start in. ;Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] OK sure then how will this associate with my register line that uses provider.com This makes no sense to me... I mean It really makes no sense... Sorry for my confusion. Do I need the register line or do I not need the register line? Why even have a register line if you don't need it and can somehow do this in a peerf, riend or user section. and if you need the register line the instructions say not to use [provider.com] as the peer, then how the heck do you get that register line to work with an associated [peer]. I need to get a handle on how this works before I go posting my sporatic attempts to get a friend,peer or user to 'register' which is not working. The only way I've been able to get my system to take incoming calls from our sip provider so far is to use register lines and keep the system 'registered' with our provider. I don't use any sip providers, so be careful with what I say here. Based on the current sip.conf.sample comments (as of today), it would appear you need to do something like this: register = 2345:[EMAIL PROTECTED]/1234 register = 2346:[EMAIL PROTECTED]/2345 register = 2347:[EMAIL PROTECTED]/3456 The above register statements are used to inform your sip provider which IP address you are coming from, and calls for each of those three accounts (eg, 2345, 2346, and 2347) will be routed to your system. In your extensions.conf, you would need something like: exten = 1234,1,Dial(SIP/3000) exten = 2345,1,Dial(SIP/3001) exten = 3456,1,Dial(SIP/3002) Note the comments in the sample config relative to not using a host= statement in the type=peer section. Also note the above register statements assume the use of three different account names (eg, 2345, 2346, and 2347). As I mentioned above, I don't use any sip providers. But, if I read the sample file correctly, the key to the above working is having three different account names. Olle has made several changes to the sip implementation in asterisk over the last year or so, so there might be variations of how this is done that are asterisk version dependent. He has also posted (several times) comments relative to how incoming sip calls match the various definitions in sip.conf. Again, since I don't use sip providers, I'll go from memory to try and repeat at least a portion of his posts. Be careful as I don't have any recent practical experience on this. It goes something like this: If you specify a host= statement in sip.conf, incoming calls will match the first section in sip.conf that includes that statement (essentially disregarding username and secret, etc). If you don't specify a host= statement in sip.conf and you have a
[Asterisk-Users] DID registration status
HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanksGiridhar Bandi ___ --Bandwidth 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 registration status
Giridhar Reddy Bandi wrote: HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this How about 'sip show registry'? ___ --Bandwidth 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 registration status
Try sip show registry from the asterisk console.Kyle On 4/2/06, Giridhar Reddy Bandi [EMAIL PROTECTED] wrote: HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanksGiridhar Bandi ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
Is the SIP phone behind NAT? That's one of the common reasons for one way audio. You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity. The SIP port ranges should be something like: SIP: 5060-5061RTP: 1-2KyleOn 4/1/06, Il Neofita [EMAIL PROTECTED] wrote: Hi,I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 2.0 Where to download
Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download.ThanksTin Trung NguyenTechnical TeamMobile: 084-91.365.4857website: www.daivietcontrol.net___ --Bandwidth 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 2.0 Where to download
On 02/04/06, Nguyen Trung Tin [EMAIL PROTECTED] wrote: Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. That version is a year and a day old now, isn't it? Peter -- Peter Bowyer Email: [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] Asterisk 2.0 Where to download
Nguyen Trung Tin wrote: Hello All I read in www.sineapps.com http://www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net Tri, That was last years April Fools joke. I still haven't heard this years... -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: caller anounce
I would also like to know how to do this, it really defeats the whole purpose of the list if you reply off list. Please post that to the list. Miles Shaun wrote: Sent you a email ~Shaun Tom Vile [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a script that will do this. Contact me off list for information. On 3/30/06, Shaun [EMAIL PROTECTED] wrote: I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the auto-attendant then tells them they have a call from plays the response from the user and gives the agent a set of options (1 to connect, 2 put on hold, 3 send to voice mail, 4 to disconnect caller). I currently have the queues setup and can get the caller connected to a agent, it's just the middle part that's missing What is the best way to achieve this, can anybody point me in the right direction? Examples are always helpful! -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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] G729 codec problems
On Sun, Apr 02, 2006 at 09:32:09AM +1000, RumaTech wrote: Sorry, I was out of action for some time. I am using Voipstnt to plae calls to USA/Canada and I bouhj 1 copy og G.729. This was mainly to get one of the local Australians VoIP providers working. Each channel needs TWO licenses, one for each way (I think). Therefore buying a single license is pretty useless. 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 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] Asterisk 2.0 Where to download
Kristian Kielhofner wrote: That was last years April Fools joke. I still haven't heard this years... And how much do you want to bet that it'll be this years 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
Re: [Asterisk-Users] FreePBX on Debian
Hi, okay, i managed to install it :) but I have some Problems with mISDN. I have set up two mISDN-extensions, that can phone each other, or the mailboxes. But i can't phone those special numbers like *60 for weather or *98 for message center, what should i do? stoffell wrote: On 4/2/06, Christian Gröger [EMAIL PROTECTED] wrote: need that Zaptel stuff? It always prompted errors so i am now using mISDN -without errors, is there a module for freePBX for mISDN? to use mISDN with freepbx, you can Add custom trunk in the Trunks menu. Anyway, is there a good manual for installing FreePBX on debian? Something with typical debian-errors and stuff? That standard manual is so focussed on Suse :( Try the readme or INSTALL file, in the archive. It basically explains what you need to do. Or join #freepbx on irc. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no audio between sip channels * 1.2.6
Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the other and no sound. Okay must have screwed something up. checked sip.conf all looked good. Okay good time to go to 1.2.6, still no audio. All phones ring and answer but no audio. the last thing that apears on the console is attempting native bridge of sip/677- and sip/699- below is a debug of a call and sip.conf. Each channel on the PAP2's is set to a different port 5060 through 5063. I can call in to any phone and all is good, use any phone to call to POTS line and back in on second POTS line and all is good. I have been looking through the archive of the mail list that I keep and have not found anything to fix my problems yet. i have transfered the registration of both PAP2's to a 1.2.0 system that I have and everything works as it should. moved 1.2.0 configs to 1.2.6 box and again no audio between sip channels. *CLI sip debug SIP Debugging enabled *CLI -- SIP read from 192.168.1.200:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: John Millican sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 143361 143361 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 16410 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941;received=192.168.1.200 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED];tag=as0767a869 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=6e91851e Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '677' -- SIP read from 192.168.1.200:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED];tag=as0767a869 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: John Millican sip:[EMAIL PROTECTED]:5060 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 0 --- (10 headers 0 lines)--- -- SIP read from 192.168.1.200:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=677,realm=asterisk,nonce=6e91851e,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=121d27cf19808e8a097930f0f969d3d7 Contact: John Millican sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 143361 143361 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 16410 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (non-NAT) Found user '677' Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.200:16410 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 699 in pap2 (domain 192.168.1.10) list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0
Re: [Asterisk-Users] no audio
--- Luis herrera [EMAIL PROTECTED] wrote: Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with phones outside my network. I call the extensions without a problem, it rings but when they answer I can't hear them and they can hear me. I set up in the SIP.CONF nat=yes I'm I missing any other setting or do I need a special switch that support asterisk. Thank you for your help. You also have to open ports on the router and forward them in. You need to forward ports 5000 (i believe) 5060,5061 and 1-2 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec problems
Steve Kennedy wrote: Each channel needs TWO licenses, one for each way (I think). Nope. The encoder/decoder licenses are counted separately, and each license you purchase entitles you to one encoder and one decoder. ___ --Bandwidth 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: caller anounce
My guess (it's only a hunch) as to why he asked for off list is for $$$. Usually when you see that the person wants $$$. I know there is a way of doing it in asterisk. My friend has it. Too lazy to get the code from his box. When I do I will post it. (I also believe that it is on the wiki). Dovid --- Miles Scruggs [EMAIL PROTECTED] wrote: I would also like to know how to do this, it really defeats the whole purpose of the list if you reply off list. Please post that to the list. Miles Shaun wrote: Sent you a email ~Shaun Tom Vile [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a script that will do this. Contact me off list for information. On 3/30/06, Shaun [EMAIL PROTECTED] wrote: I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the auto-attendant then tells them they have a call from plays the response from the user and gives the agent a set of options (1 to connect, 2 put on hold, 3 send to voice mail, 4 to disconnect caller). I currently have the queues setup and can get the caller connected to a agent, it's just the middle part that's missing What is the best way to achieve this, can anybody point me in the right direction? Examples are always helpful! -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: caller anounce
I have searched high and low on the wiki, the problem is that it can fit under a number of different names, since this process can be used for tons of different applications. If you find it let me know, I really want this info, as it is the last piece of the puzzle for my asterisk box. Thanks Miles Dovid Bender wrote: My guess (it's only a hunch) as to why he asked for off list is for $$$. Usually when you see that the person wants $$$. I know there is a way of doing it in asterisk. My friend has it. Too lazy to get the code from his box. When I do I will post it. (I also believe that it is on the wiki). Dovid --- Miles Scruggs [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
Re: [Asterisk-Users] Asterisk 2.0 Where to download
Your just one day too late. Some one did the same last year. I have the email some where on my server. Goto be on the game and original. --- Nguyen Trung Tin [EMAIL PROTECTED] wrote: Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap channels - help
Hi Tzafrir, thank´s for your help.
My configurations:
#zaptel.confspan=1,2,0,cas,hdb3cas=1-31:1101loadzone=usdefaultzone=us
span=2,1,0,ccs,hdb3bchan=32-46dchan=47bchan=48-62loadzone=usdefaultzone=us
#zapata.conf[trunkgroup]
[channels]context=defaultswitchtype=euroisdnsignalling=pri_net;rxwink=300usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yes
threewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=2callgroup=2immediate=nocallerid=asreceivedmusiconhold=default
group=2channel=32-46channel=48-62
2006/4/1, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Mar 31, 2006 at 10:36:02PM -0300, Josué Conti wrote: I am installing one asterisk, to establish connection with my PABX Siemens,
in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the commandexten = _ 19, 1,
dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial(SIP/8110-a729, zap/g2/1971411234|30) in new stack -- Called g2/1971411234 -- Channel 0/1, span 2 got hangup
-- Hungup 'Zap/32-1' == No one is available to answer at this timeHowever, when use the rule exten = _ 7xxx, 1, dial(zap/g2/${EXTEN}, 30) I obtain to call the branches pabx, normally.
-- Executing Dial(SIP/8110-71ee, zap/g2/7500|30) in new stack -- Called g2/7500 -- Zap/32-1 is ringing -- Zap/32-1 answered SIP/8110-71ee -- Channel 0/1, span 2 got hangup
-- Hungup 'Zap/32-1' == Spawn extension (default, 7500, 1) exited non-zero on 'SIP/8110-71ee' Somebody would have some idea to help in this case me? Greatings JosuéCould you please post your
zaptel.conf and zapata.conf ?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] Asterisk hosted solution
--- Thorben Jensen [EMAIL PROTECTED] wrote: http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk Whats the catch ? When do you stop giving it for free since these people are on your system they are forced to now pay. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Blacklist out bound numbers from file
You can do the following. Create a file called blacklist.conf . In it create the following. [BlackList] Exten = 5551212,1,Playback(num-blacklist) Exten = 5551212,2,Hangup Exten = 1900.,1,Playback(num-blacklist) Exten = 1900.,2,Hnagup You get the idea. In your extensions.conf put in the top #include /directory/of/file/blacklist.conf and then in the context place include = BlackList It should work. Dovid --- Jeremy [EMAIL PROTECTED] wrote: The lookupblacklist cmd, looks to me like it only handles inbound numbers. Can I use this command just to block outgoing numbers, by where I place it in my outoging call settings? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Melcon Moraes Sent: Wednesday, March 29, 2006 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Blacklist out bound numbers from file Try the LookupBlacklist application. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupBlacklist []'s MM -Original Message- From: Jeremy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: Sent: Wed, 29 Mar 2006 20:11:09 -0500 Delivered: Wed, 29 Mar 2006 19:13:45 Subject:[Asterisk-Users] Blacklist out bound numbers from file I'm looking to bock a list of numbers users cant call. Is it possible to pull these from file specified in the dial plan, as apposed to mysql? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,114 3681226.265135.4834.arrino.terra.com.br,3927,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 2.0 Where to download
Think people will fall for it again next year too? Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 2.0 Where to download
Think people will fall for it again next year too? Newbies will __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming triggers seperate outbound
I will post this info to the list when I get a chance. --- Miles Scruggs [EMAIL PROTECTED] wrote: Hey, I would like in the course of dial plan logic, to trigger a separate outbound call. If that outbound call is answered, and if that certain key response is detected then it will bridge the incoming call to the newly dialed outbound call. What I want to accomplish is that when a caller dials in, they can enter enter an extension that will call out to a callee's cell phone. When the callee answers their cell they have to dial 111 or some other combo to accept the call. when this is done only then will the two calls be connected. Thanks Miles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 nat problems.
I dont have that option in my phone, this is software 8.2 (p003-08-02 if i remember correctly) ~Shaun Doug Lytle [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Shaun wrote: I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone directly into the cable modem and now the phone can connect fine and works. I pasted below the sip debug output, anybody know what's going on or have experience with this? You need to turn on the NAT support for the 7960 on the phone itself. Settings -- SIP Configuration --NAT Enabled (Y/N) 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 ___ --Bandwidth 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: Cisco 7960 nat problems.
Dont have that option, i think i remember seeing it in version 7.xx but i cant remember, i may have to downgrade it i guess ~Shaun Rich Adamson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Shaun wrote: I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone directly into the cable modem and now the phone can connect fine and works. I pasted below the sip debug output, anybody know what's going on or have experience with this? I have several of these working just fine. You really do need nat=yes in sip.conf, and, change the 7960 config to say nat=yes in the 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 ___ --Bandwidth 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] can automon work with MixMonitor
Hi list automon now works as Monitor does. But MixMonitor is a better way for most cases, I guess. Any workaround to make automon do that? Any help will be appreciated. Franz Wu ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk answering machine replacement, WaitForRing(), application return values
This is possibly a dumb question, but I've googled around and poked through the documentation and I'm a bit confused. My initial experiment with Asterisk involves setting it up in place of my old dedicated answering machine. That means I've still got a regular old phone on the line which we normally answer calls with.[1] I'm trying to set up Asterisk to leave the line alone if someone picks up the regular phone within a certain number of rings. Google came up with a few people asking how to do the same thing but no definitive descriptions or indications as to what worked. I THINK I actually have this set up now, but I'm unclear on how WaitForRing() is supposed to work. Depending on which set of documentation you look it, it seems WaitForRing($seconds) is supposed to either A)Wait $seconds seconds AFTER the next ring it detects or B)Wait $seconds for the next ring (and fail if no ring occurs in that time) or possibly C)Wait $seconds seconds, then listen for the next ring. (The OReilly book has one description for the application as the summary, then another as the full description...) I'm operating on the assumption that B) is supposed to be correct when you use WaitForRing() in the context of an incoming call? Also, while googling I did find that you can't do anything with the return values of applications (looks like the return values are only for programmers writing asterisk modules). Is it correct that if an applications fails (returns -1) the thread stops at that point due to the failure? Basically, my approach in extensions.conf right now is: ;wait about 3-4 rings Wait(14) ;listen for one more ring - if there isn't one, leave the line alone ;because we assume someone picked up the regular phone in time WaitForRing(1) ;I assume if WaitForRing(1) fails [returns -1] it kills the ;thread automatically? ;otherwise - answer and have Asterisk take over handling the call Answer() etc. (Any other recommended approaches for this would be appreciated, too...) Thanks, all [1] money is an issue, otherwise I could just swap in a VoIP phone or buy the expensive (for my budget at the moment) TDM card with an FXO and FXS module instead of the cheap X101P clone I'm using now. pgpPTYn4dLNHJ.pgp Description: 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
[Asterisk-Users] Subversion mirrors of Asterisk, Zaptel and libpri rebuilt
Due to an error in the configuration of the mirroring tool we are using to mirror the repositories from our internal commit server to the public read-only mirror, the revision numbers were not being properly kept in sync (so rev 14381 on the internal server was not the same as on the mirror). This problem has just been corrected, but it required rebuilding the repositories on the mirror server from scratch. If you have existing SVN checkouts of these repositories, you will probably not be able to 'svn update' them any longer, as the repository ID is different and the revision numbers don't match up. Sorry for the inconvenience, but fixing this problem was necessary... ___ --Bandwidth 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: Cisco 7960 nat problems.
Shaun Reitan wrote: I dont have that option in my phone, this is software 8.2 (p003-08-02 if i remember correctly) That is the firmware that I am also running. It's under SIP Configuration, all the way at the bottom. Not within the extension confgs, keep hitting the down arrow until you get to option 24. 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] Connecting Asterisk to traditional phone central
Hi,I am trying to connect Asterisk to traditional central. It must be done over ISDN. I am utilizing Asterisk 1.0.9 from bristuff RC8p. I was able to communicate them with each other, but I have a problem with callerid. The traditional central does not recognize the callerid from the phones connected to Asterisk over Linksys PAP-2. The service man from the traditional central told me that in logs of the central the callerid is not set. To check it I connected the normal ISDN phone to the central and in this case the callerid was set. I think the the problem is in my Asterisk box. My zapata.conf looks like this:; Zapata telephony interface; Configuration file[channels]switchtype = euroisdnsignalling = bri_cpe_ptmppridialplan = local nationalprefix = 0internationalprefix = 00echocancel = yesthreewaycalling = yestransfer = yesimmediate = nocallerid = asreceived usecallerid = yesgroup = 1context = isdnchannel = 1-2Does anyone have any tips or tricks?CheersAndrew ___ --Bandwidth 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] Newbie question - sip.conf incoming contexts
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension needs
to
; be defined in extensions.conf to be able to accept calls from this SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;connect to local extension 1234 in extensions.conf, default context,
;unless you configure a [sip_proxy] section below, and configure a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm already
in using a register line
the calls some into the context specified in [general] section of sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Sorry for my confusion.
Do I need the register line or do I not need the register line?
Why even have a register line if you don't need it and can somehow
do this in a peerf, riend or user section.
and if you need the register line the instructions say
not to use [provider.com] as the peer, then how the heck do you
get that register line to work with an associated [peer].
I need to get a handle on how this works before I go posting my
sporatic attempts to get a friend,peer or user to 'register'
which is not working.
The only way I've been able to get my system to take incoming calls
from our sip provider so far is to use register lines and keep
the system 'registered' with our provider.
I don't use any sip providers, so be careful with what I say here.
Based on the current sip.conf.sample comments (as of today), it would
appear you need to do something like this:
register = 2345:[EMAIL PROTECTED]/1234
register = 2346:[EMAIL PROTECTED]/2345
register = 2347:[EMAIL PROTECTED]/3456
The above register statements are used to inform your sip provider which
IP address you are coming from, and calls for each of those three
accounts (eg, 2345, 2346, and 2347) will be routed to your system. In
your extensions.conf, you would need something like:
exten = 1234,1,Dial(SIP/3000)
exten = 2345,1,Dial(SIP/3001)
exten = 3456,1,Dial(SIP/3002)
Note the comments in the sample config relative to not using a host=
statement in the type=peer section. Also note the above register
statements assume the use of three different account names (eg, 2345,
2346, and
[Asterisk-Users] Voicemail() - Reading exit or return results
Here my script:
exten = 230,1,Answer exten
= 230,2,NoOpexten =
230,3,Voicemail(u${EXTEN})exten = 230,4,NoOp(Need results
from VoiceMail() above - should be non-zero and provided to ARG3)
exten = 230,5,NoOpexten =
230,6,GoToIf($[${ARG3} = 0]?s|8) exten =
230,7,system(/var/lib/asterisk/agi-bin/230.php|${EXTEN}) ; exten
= 230,8,Hangup
I need to know how to read and use 'exit' results
into ARG3 from voicemail() so the script will continue past line 6 or
not
Thanks
Bart
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RE: [Asterisk-Users] Connecting Asterisk to traditional phone central
Use a set callerid before your dial statement in extensions.conf. -Original Message- From: Andrew Nowrot [mailto:[EMAIL PROTECTED] Sent: Sun 4/2/2006 4:19 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Connecting Asterisk to traditional phone central Hi, I am trying to connect Asterisk to traditional central. It must be done over ISDN. I am utilizing Asterisk 1.0.9 from bristuff RC8p. I was able to communicate them with each other, but I have a problem with callerid. The traditional central does not recognize the callerid from the phones connected to Asterisk over Linksys PAP-2. The service man from the traditional central told me that in logs of the central the callerid is not set. To check it I connected the normal ISDN phone to the central and in this case the callerid was set. I think the the problem is in my Asterisk box. My zapata.conf looks like this: ; Zapata telephony interface ; Configuration file [channels] switchtype= euroisdn signalling = bri_cpe_ptmp pridialplan = local nationalprefix = 0 internationalprefix = 00 echocancel = yes threewaycalling = yes transfer = yes immediate = no callerid = asreceived usecallerid= yes group = 1 context= isdn channel = 1-2 Does anyone have any tips or tricks? Cheers Andrew winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec problems
Hi, I wonder if VoIP providers consume two licenses when one calls via them? One license for my phone to the provider and one license when call is passed to the recepient. Is that possible? Rudolf On 4/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Steve Kennedy wrote: Each channel needs TWO licenses, one for each way (I think). Nope. The encoder/decoder licenses are counted separately, and each license you purchase entitles you to one encoder and one decoder. ___ --Bandwidth 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] Codec Problem
I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this errorAsked to transmit frame type 256, while native formats is 4 (read/write = 4/4) ___ --Bandwidth 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] Information about LOCAL/ Channel
Hi, I have searched the web and found some basic information about the LOCAL/ channel. I am wondering if anybody has any good web resources they can point me to on this subject as I think I will need to use it in the near future in a solution I am putting together. Alternatively, if anybody has a dial plan where they have used the LOCAL/ channel and they are willing to let me have a look at it that would also be greatly appreciated as I could imagine I can work backward from there. Kind Regards Stuart begin:vcard fn:Stuart Elvish n:Elvish;Stuart org:Dallas Delta Corporation Pty Ltd;Voice Networking Directorate adr:;;102 Albert Street;East Brunswick;VIC;3057;Australia email;internet:[EMAIL PROTECTED] title:Voice Networking Engineer tel;work:03 9387 7445 tel;fax:03 9387 3128 tel;cell:0408 873 601 x-mozilla-html:TRUE url:http://www.dallasdelta.net version:2.1 end:vcard ___ --Bandwidth 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] Information about LOCAL/ Channel
Yes I have used it and it looks like this:
exten = s,1,Dial(Local/[EMAIL PROTECTED])
exten = s,2,Goto(s-${DIALSTATUS},1)
On 4/2/06, Stuart Elvish - Dallas Delta Corporation Pty Ltd
[EMAIL PROTECTED] wrote:
Hi,
I have searched the web and found some basic information about the
LOCAL/ channel. I am wondering if anybody has any good web resources
they can point me to on this subject as I think I will need to use it in
the near future in a solution I am putting together.
Alternatively, if anybody has a dial plan where they have used the
LOCAL/ channel and they are willing to let me have a look at it that
would also be greatly appreciated as I could imagine I can work backward
from there.
Kind Regards
Stuart
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Re: [Asterisk-Users] Re: caller anounce
Actually you are wrong about $$$ Dovid, I did not charge one penny for it. On 4/2/06, Dovid Bender [EMAIL PROTECTED] wrote: My guess (it's only a hunch) as to why he asked for off list is for $$$. Usually when you see that the person wants $$$. I know there is a way of doing it in asterisk. My friend has it. Too lazy to get the code from his box. When I do I will post it. (I also believe that it is on the wiki). Dovid --- Miles Scruggs [EMAIL PROTECTED] wrote: I would also like to know how to do this, it really defeats the whole purpose of the list if you reply off list. Please post that to the list. Miles Shaun wrote: Sent you a email ~Shaun Tom Vile [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a script that will do this. Contact me off list for information. On 3/30/06, Shaun [EMAIL PROTECTED] wrote: I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the auto-attendant then tells them they have a call from plays the response from the user and gives the agent a set of options (1 to connect, 2 put on hold, 3 send to voice mail, 4 to disconnect caller). I currently have the queues setup and can get the caller connected to a agent, it's just the middle part that's missing What is the best way to achieve this, can anybody point me in the right direction? Examples are always helpful! -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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: caller anounce
There is a reason why I am posting it off list and not because of money. On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote: I would also like to know how to do this, it really defeats the whole purpose of the list if you reply off list. Please post that to the list. Miles Shaun wrote: Sent you a email ~Shaun Tom Vile [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a script that will do this. Contact me off list for information. On 3/30/06, Shaun [EMAIL PROTECTED] wrote: I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the auto-attendant then tells them they have a call from plays the response from the user and gives the agent a set of options (1 to connect, 2 put on hold, 3 send to voice mail, 4 to disconnect caller). I currently have the queues setup and can get the caller connected to a agent, it's just the middle part that's missing What is the best way to achieve this, can anybody point me in the right direction? Examples are always helpful! -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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: caller anounce
um ok, well do you mind posting it off list to myself, if you haven't caught it I am interested. Miles Tom Vile wrote: There is a reason why I am posting it off list and not because of money. On 4/2/06, Miles Scruggs [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
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1) ;
exten = 5185551234,1,SetVar(FROM_DID=5185551234) ;
exten = 5185551234,2,Goto(custom-callid,s,1) ;
On 4/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
needs to
; be defined in extensions.conf to be able to accept calls from this SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;connect to local extension 1234 in extensions.conf, default context,
;unless you configure a [sip_proxy] section below, and configure a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm already
in using a register line
the calls some into the context specified in [general] section of sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Sorry for my confusion.
Do I need the register line or do I not need the register line?
Why even have a register line if you don't need it and can somehow
do this in a peerf, riend or user section.
and if you need the register line the instructions say
not to use [provider.com] as the peer, then how the heck do you
get that register line to work with an associated [peer].
I need to get a handle on how this works before I go posting my
sporatic attempts to get a friend,peer or user to 'register'
which is not working.
The only way I've been able to get my system to take incoming calls
from our sip provider so far is to use register lines and keep
the system 'registered' with our provider.
I don't use any sip providers, so be careful with what I say here.
Based on the current sip.conf.sample comments (as of today), it would
appear you need to do something like this:
register = 2345:[EMAIL PROTECTED]/1234
register = 2346:[EMAIL PROTECTED]/2345
register = 2347:[EMAIL PROTECTED]/3456
The above register statements are used to inform your sip provider which
IP address you
Re: [Asterisk-Users] Re: caller anounce
um ok, maybe not since you seemed a bit rude. On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote: um ok, well do you mind posting it off list to myself, if you haven't caught it I am interested. Miles Tom Vile wrote: There is a reason why I am posting it off list and not because of money. On 4/2/06, Miles Scruggs [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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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] Reporting?
Nicolas, Do you have any idea what this will cost and when it might be released? Thanks, Doug -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nicolás Gudiño Sent: Friday, March 31, 2006 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Reporting? NICE! On 3/30/06, Joe Dennick [EMAIL PROTECTED] wrote: I see (and like) the demo, but where can we get it? Doug Lytle wrote: Nicolás Gudiño wrote: shameless plug Something like this perhaps? http://www.asternic.org/stats/demo It is not released yet... I'm not having much time to write the web page, documentation, tarball, etc. -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth 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] Who is on a call?
I would like to know which extension number is engaged in a call. show channels shows me: *CLI show channels Channel Location State Application(Data) SIP/asterisk.elmit.com-0 [EMAIL PROTECTED]:2Up Echo() SIP/8807-066 [EMAIL PROTECTED] Up Echo() 2 active channels 2 active calls but it is not true!!! show channels verbose gives me even a time to each of 112:43:33 and 347:23:22 I want to know: 1. is extension number 444 in use (calls) 2. is the connection to my provider abc in use (call) How can I get this info as CLI comand and as a jump criteria in the dialplan bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: caller anounce
eh? what have I done that is rude, I never even made comments about money? Tom Vile wrote: um ok, maybe not since you seemed a bit rude. On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote: um ok, well do you mind posting it off list to myself, if you haven't caught it I am interested. Miles Tom Vile wrote: There is a reason why I am posting it off list and not because of money. On 4/2/06, Miles Scruggs [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
Re: [Asterisk-Users] Re: caller anounce
OK, enough of this.. No reason to bicker about something like this.
Here is the URL.
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dialdiff=57
For those of you that do not have a working web browser or cand find it with
Google here is the text.
Dial macros
Introduced in/for Asterisk 1.2, see bug/patch 2905
You can now add args to the macro by using a '^' char
Dial(Zap/1|60|M(mymacro^cat^dog^bark))
Also, the macro can set the MACRO_RESULT variable to do the following:
ABORT - Hangup both legs of the call
BUSY
CONTINUE - Hangup the called party and continue on in the dialplan from
where you called Dial
GOTO:context^exten^priority - Transfer the call.
Note: If you want the call to be bridged upon completion of the macro, you
should NOT set the MACRO_RESULT variable to anything. IF MACRO_RESULT is not
defined, the thread of execution falls off the end of the macro and bridges
the call. Setting it to CONTINUE causes the call NOT to be bridged, and
execution to resume at n+1 priority in the calling context. Of course
setting it to BUSY or GOTO has the implied results (and of course the call
is not bridged).
(I found this a bit confusing because my mental model expected some kind of
explicit case for bridge the call, such as setting MACRO_RESULT to
BRIDGE or CONNECT or something like that. It's the ABSENCE of any value
that causes the call to be bridged.)
Example 2: Dial macro
screen-record: Please record your name press pound when finished.
screen-from: You have a call from
screen-accept: Press 1 to accept this call or any other key to reject.
exten = 890,1,Wait(0.2)
exten = 890,2,Playback(screen-record)
exten = 890,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = 890,4,Record(${SCREEN_FILE}.gsm|6|25)
exten = 890,5,Dial(SIP/16|60|gM(screen^${SCREEN_FILE}))
exten = 890,6,Voicemail([EMAIL PROTECTED])
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(screen-from)
exten = s,3,Playback(${ARG1})
exten = s,4,Read(ACCEPT|screen-accept|1)
exten = s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten = s,6,SetVar(MACRO_RESULT=CONTINUE)
exten = s,7,System(/bin/rm ${ARG1})
Notes:
a.. Do not put spaces between the arguments to the Dial command, it will
not work.
b.. When options t, T, h, H, w, W or L (with multiple
arguments) are applied, Asterisk will remain in the media path, even if
canreinvite=yes'' (a SIP channel option) has been specified.
Return codes
Dial sets DIALSTATUS to indicate its success. However, under some
circumstances, execution will jump to priority n+101 in the current context.
This happens when:
a.. All channels dialed were busy
b.. There is exists something at n+101 in the current context
c.. You are running asterisk 1.0.x, priorityjumping=yes is set in
extensions.conf, or the j option is specificed in the dial command
Note that in Asterisk v1.2+ priorityjumping is considered off by default,
which is a change from previous versions.
If the g option is specified, and the called party hangs up before the
calling party, then Dial continues execution at priority n+1.
Note: The bristuff patches change the dial behaviour slightly and jump to
n+201 if the dialed destination isn't connected (e.g. a SIP softphone is not
up and running, or hasn't registered).
___
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[Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma
Hello List! I wanted to share to everyone the following compatible connectivity products that my company installed in our Asterisk based soft switch. I already sent these to the Asterisk.org site many days ago but for some reason they still have to post it. 1. Sangoma A101 single port E1/T1/PRI Card 2. Sangoma A102 dual port E1/T1/PRI Card 3. Sangoma A104 quad port E1/T1/PRI Card 4. Sangoma A104D quad port E1/T1/PRI Card 5. Sangoma A200 FXS/FXO Series Above products will work with any commercially available motherboard in the market. This we have proven true because we have at least 5 different servers that we have installed them to. And take note of their on-board echo cancellers, simply works superb! I hope the above helps especially to those who're just starting with Asterisks. Can anyone from Asterisk.Org please have the above posted? Thanks. Heidi __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapBarge but ability to talk to the agent
Hi, Is there a way to do a ZapBarge, but where the person doing the barge-in would be able to talk to the agent only (whispering)? Thanks, Andre Courchesne - Consultant http://www.net-forces.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] H323 on way voice
Neofita wrote: -- channelsOpen = 1 There is only ONE channel open. This should be a huge alarm to you. Jeremy McNamara ___ --Bandwidth 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] Who is on a call?
The 'sip show channels' and 'show channels' command aren't exactly easy to interpret, especially if one of the numbers has pic codes and rate centers inserted (the rest is truncated on the output), or you have a proxy involved in the call. Wish someone with some C knowledge would fix that. Doug. -Original Message- From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] Sent: Sun 4/2/2006 8:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] Who is on a call? I would like to know which extension number is engaged in a call. show channels shows me: *CLI show channels Channel Location State Application(Data) SIP/asterisk.elmit.com-0 [EMAIL PROTECTED]:2Up Echo() SIP/8807-066 [EMAIL PROTECTED] Up Echo() 2 active channels 2 active calls but it is not true!!! show channels verbose gives me even a time to each of 112:43:33 and 347:23:22 I want to know: 1. is extension number 444 in use (calls) 2. is the connection to my provider abc in use (call) How can I get this info as CLI comand and as a jump criteria in the dialplan bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] H323 on way voice
[EMAIL PROTECTED] wrote: I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. So instead of solving his configuration problem he should try a new channel driver? Michael Mansos(or something like that) and other guys have been done a good job. And myself and the others that have contributed to chan_h323 work haven't? Get a life - Think before you type. Jeremy McNamara ___ --Bandwidth 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: caller anounce
Great thanks
Steven Job wrote:
OK, enough of this.. No reason to bicker about something like
this.
Here is the URL.
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dialdiff=57
For those of you that do not have a working web browser or cand find
it with Google here is the text.
Dial macros
Introduced in/for Asterisk 1.2, see bug/patch 2905
You can now add args to the macro by using a '^' char
Dial(Zap/1|60|M(mymacro^cat^dog^bark))
Also, the macro can set the MACRO_RESULT variable to do the following:
ABORT - Hangup both legs of the call
BUSY
CONTINUE - Hangup the called party and continue on in the dialplan
from where you called Dial
GOTO:context^exten^priority - Transfer the call.
Note: If you want the call to be bridged upon completion of the macro,
you should NOT set the MACRO_RESULT variable to anything. IF
MACRO_RESULT is not defined, the thread of execution falls off the end
of the macro and bridges the call. Setting it to CONTINUE causes the
call NOT to be bridged, and execution to resume at n+1 priority in the
calling context. Of course setting it to BUSY or GOTO has the implied
results (and of course the call is not bridged).
(I found this a bit confusing because my mental model expected some
kind of explicit case for bridge the call, such as setting
MACRO_RESULT to BRIDGE or CONNECT or something like that. It's the
ABSENCE of any value that causes the call to be bridged.)
Example 2: Dial macro
screen-record: Please record your name press pound when finished.
screen-from: You have a call from
screen-accept: Press 1 to accept this call or any other key to reject.
exten = 890,1,Wait(0.2)
exten = 890,2,Playback(screen-record)
exten = 890,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = 890,4,Record(${SCREEN_FILE}.gsm|6|25)
exten = 890,5,Dial(SIP/16|60|gM(screen^${SCREEN_FILE}))
exten = 890,6,Voicemail([EMAIL PROTECTED])
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(screen-from)
exten = s,3,Playback(${ARG1})
exten = s,4,Read(ACCEPT|screen-accept|1)
exten = s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten = s,6,SetVar(MACRO_RESULT=CONTINUE)
exten = s,7,System(/bin/rm ${ARG1})
Notes:
a.. Do not put spaces between the arguments to the Dial command, it
will not work.
b.. When options t, T, h, H, w, W or L (with multiple
arguments) are applied, Asterisk will remain in the media path, even
if canreinvite=yes'' (a SIP channel option) has been specified.
Return codes
Dial sets DIALSTATUS to indicate its success. However, under some
circumstances, execution will jump to priority n+101 in the current
context. This happens when:
a.. All channels dialed were busy
b.. There is exists something at n+101 in the current context
c.. You are running asterisk 1.0.x, priorityjumping=yes is set in
extensions.conf, or the j option is specificed in the dial command
Note that in Asterisk v1.2+ priorityjumping is considered off by
default, which is a change from previous versions.
If the g option is specified, and the called party hangs up before the
calling party, then Dial continues execution at priority n+1.
Note: The bristuff patches change the dial behaviour slightly and jump
to n+201 if the dialed destination isn't connected (e.g. a SIP
softphone is not up and running, or hasn't registered).
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
___
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[Asterisk-Users] ASTCC: How to reset in-use flag automatically ?
I have some troubles with ASTCC. TO often the in-use flag remains set. I would like to find a solution, where astcc.agi checks automatically if THIS user is in a call rather than to check the flag. If that is not possible, I would like to have an extension to dial to, and it will after hang up, reset the flag! The in-use flag remains set, if the caller hang up before the gateway gets the call. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
