Re: [asterisk-users] Asterisk version survey
As I said before. I would like to see a much more comprehensive survey complete with names or nicknames if supplied. Asterisk version, in-house patches or programming, OS, preferred hardware (server, phones, cards) Then more bio on the respondent. Years in data and also telephony field, other phone systems certified or capable of deploying or servicing. Number of installs, number of stations/trunks, function of deployments. Maybe even an breakdown on age group as well as general comments. If $9 can put that survey together in a comprehensible set of questions and results, I will pay the $9. Thanks, Steve 888.777.1888 randulo wrote: Yes, unless I pay $9, it's limited to 100 responses. The clear winner was debian, well after other. My fault for not limiting to multiple choice, but reading through the 100 comments is interesting. Maybe I can get Digium to kick in the money for a full survey :) regards, r On Nov 27, 2007 7:58 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Your form can no longer accept submissions. SuSe 10.1 with latest Asterisk 1.2 using our own patches. We are about ready to go live with new installations of SLES or CentOS + Asterisk 1.4 just need to work out the bugs. On Nov 26, 2007 5:14 AM, randulo [EMAIL PROTECTED] wrote: Hi, I'd like to invite all asterisk users to answer two questions on this form: http://food4wine.ning.com/poll 1) What version do you use in production (1.2, 1.4 or both) 2) and what distro(s) It'll just take a second and the results are public and live (link on the page above) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
Well, $9 would pay for up to 500 answers. I also found a free one I'm looking at now, but you never get anything really good free :) If $9 can put that survey together in a comprehensible set of questions and results, I will pay the $9. Let me see if I can put what you ask for together on the free one and post here in a bit. r ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
NEW Asterisk favorite OS survey: http://www.esurveyspro.com/Survey.aspx?id=1f38482e-c3fa-4384-8b8a-65a9f40b2cd8 The above is a compromise between three elements: 1) The amount of time and patience I have 2) The information I think most people are willing to proovide 3) Steve's requests Please have at it. Steve, you can use this same setup free to design your own. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
randulo wrote: Well, $9 would pay for up to 500 answers. I also found a free one I'm looking at now, but you never get anything really good free :) If $9 can put that survey together in a comprehensible set of questions and results, I will pay the $9. Let me see if I can put what you ask for together on the free one and post here in a bit. r Can you just install limesurvey on a server some place? It would allow you to do however many future surveys you want to do. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Julio, Thanks for your suggestion, at this stage I would like this version of g729 running in my box.. but,,, is good to know the paid version works without any problems in this machine for the next stage. Best Regards, Fernando Julio Arruda wrote: Fernando Berretta wrote: Dear Mindaugas, Thanks for your promt response I've already tried this but.. it's not working,, what file do you think I should use ? any other idea ? Fernando, I've used the official/legal G729 codec sold at www.digium.com in Athlon boxes w/ asterisk 1.4 without problems, have you tried this option ? Mindaugas Kezys wrote: Rename to codec_g729.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so Copy to /usr/lib/asterisk/modules chmod 777 codec_g729.so restart Asterisk show translations Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Fernando Berretta *Sent:* Monday, November 26, 2007 6:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Dear Mindaugas, I've already download the folowing files for testing codec_g729-ast14-gcc4-glibc-athlon-sse.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-icc-glibc-x86_64-core2.so But... no one of them seems to be working ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk behind a PIX firewall?
Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 1-2 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 1-2.The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra 9133i. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
y the way, CentOS is winning at the moment. I don't see how to publish the results live yet, but I'll do a static report as soon as there are more. On Nov 27, 2007 2:25 PM, randulo [EMAIL PROTECTED] wrote: On Nov 27, 2007 2:06 PM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Can you just install limesurvey on a server some place? It would allow you to do however many future surveys you want to do. I'll look into it for my own use, thanks. Wufoo is way ahead of most of these, but it ain't free, and I understand why. If I need a paid survey/poll tool I'll use them. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
On Nov 27, 2007 2:06 PM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Can you just install limesurvey on a server some place? It would allow you to do however many future surveys you want to do. I'll look into it for my own use, thanks. Wufoo is way ahead of most of these, but it ain't free, and I understand why. If I need a paid survey/poll tool I'll use them. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt wrote: Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 1-2 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 1-2. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra 9133i. Just checking NAT=yes, canreinvite=no ? Thanks, Steve Totaro 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip to ATA?
Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding how the phone service works. All I can get is I plug my existing phones and answering machines into the back of the cable modem and am good to go. I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP port 5060 closed - how do I open it?
Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0 allowexternaldomains=no allowexternalinvites=no Do I have to set allowexternalinvites or allowexternaldomains to yes to accept INVITEs from my ITSP? I've already configured the system to allow traffic from their IP address. Thanks for the help! Regards, Zaheer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I live OpenVPN bridges for double NAT situations, of course you could try IAX2 but I have seen too many sound quality issues surrounding IAX2 so I try to stick with SIP, even if that means setting up VPNs. Thanks, Steve 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk API Manager
Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I live OpenVPN bridges for double NAT situations, of course you could try IAX2 but I have seen too many sound quality issues surrounding IAX2 so I try to stick with SIP, even if that means setting up VPNs. This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt wrote: Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I live OpenVPN bridges for double NAT situations, of course you could try IAX2 but I have seen too many sound quality issues surrounding IAX2 so I try to stick with SIP, even if that means setting up VPNs. This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? It is being tunneled or forwarded? Does the Asterisk box have a public IP or does the PIX have the public which just forwards to the private? If it is just forwarding, it will never work without either putting one side on a public IP, using a VPN solution, or IAX2. Thanks, Steve 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? If I remove the phone from behind the Netgear... then I get the audio from the Asterisk PBX so traffic seems to be flowing but why would it not get behind the firewalls? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
On Nov 27, 2007 9:59 AM, Matt [EMAIL PROTECTED] wrote: This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? If I remove the phone from behind the Netgear... then I get the audio from the Asterisk PBX so traffic seems to be flowing but why would it not get behind the firewalls? This is what I see on the debug: etransmitting #6 (NAT) to 63.174.244.147:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 63.174.244.147;branch=z9hG4bK7e4d50af2;received= 63.174.244.147 From: Remote Test sip:[EMAIL PROTECTED]:5060;tag=c302787b4625316 To: 93372806 sip:[EMAIL PROTECTED]:5060;tag=as1c9e4806 Call-ID: [EMAIL PROTECTED] CSeq: 1136993892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 242 The From and To shouldn't be the same, though... should they? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt wrote: This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? If I remove the phone from behind the Netgear... then I get the audio from the Asterisk PBX so traffic seems to be flowing but why would it not get behind the firewalls? Trust me on this, I have tried almost everything to get it to work, the best you can hope for is one way audio in a dual NAT. The answer has to do with where the packets are sent from and where they seem to be sent from. If you are not familiar with OpenVPN, you should check it out. It is a great piece of software and will solve your issues. Thanks, Steve Totaro 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
It is being tunneled or forwarded? Does the Asterisk box have a public IP or does the PIX have the public which just forwards to the private? If it is just forwarding, it will never work without either putting one side on a public IP, using a VPN solution, or IAX2. It IS being forwarded. Asterisk has a private, and the PIX forwards... and I do see what is happening. Makes sense. Guess it's going to have to run over the VPN! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt, If your phone is using SIP, then you should enable sip inspection (7.x code or above) or fixup sip (6.x code) and have a rule that allows source (wherever you need) inbound on the outside interface to TCP 5060 (SIP port). The sip inspection or fixup should enable the proper ports for the require RTP streams. I had this working through an ASA at some point, but I don't remember if both ends were doing NAT or only one end. I don't know the phone you are talking about, but you also might want to look into STUN or ICE to get beyond the NAT Traversal issue, if that is what's causing the problem. In the Firewall log, are you seeing Denys? or drops? Have you tried debug sip on the firewall console? I've been dealing with several ASA SIP issues lately. SIP trunking with NAT will certainly not work and there is a Cisco Bug that my company discovered when setting up our PBX. Shlomo in Israel On 11/27/07, Matt [EMAIL PROTECTED] wrote: Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 1-2 UDP are open on the PIX and forwarding to the Asteriskserver. The Asterisk server's RTP.CONF is set to use 1-2.The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra 9133i. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Trust me on this, I have tried almost everything to get it to work, the best you can hope for is one way audio in a dual NAT. The answer has to do with where the packets are sent from and where they seem to be sent from. If you are not familiar with OpenVPN, you should check it out. It is a great piece of software and will solve your issues. Steve, Thanks for the informationI guess we will go with VPN. A little Sokris board isn't that expensive to throw at each site. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Zaheer, If a netstat -an|grep -I LISTENING shows that a LISTENING port for 5060 is there, then the problem isn't Asterisk, but some firewall system on the server is blocking access from outside. If its not there, then come back to the group.. Adrian Marsh From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: 27 November 2007 14:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP port 5060 closed - how do I open it? Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0 allowexternaldomains=no allowexternalinvites=no Do I have to set allowexternalinvites or allowexternaldomains to yes to accept INVITEs from my ITSP? I've already configured the system to allow traffic from their IP address. Thanks for the help! Regards, Zaheer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Correction: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* Adrian Marsh From: Adrian Marsh Sent: 27 November 2007 15:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] SIP port 5060 closed - how do I open it? Zaheer, If a netstat -an|grep -I LISTENING shows that a LISTENING port for 5060 is there, then the problem isn't Asterisk, but some firewall system on the server is blocking access from outside. If its not there, then come back to the group.. Adrian Marsh From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: 27 November 2007 14:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP port 5060 closed - how do I open it? Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0 allowexternaldomains=no allowexternalinvites=no Do I have to set allowexternalinvites or allowexternaldomains to yes to accept INVITEs from my ITSP? I've already configured the system to allow traffic from their IP address. Thanks for the help! Regards, Zaheer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Try to just open port 5060 for SIP signaling on the PIX and also enable the INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and open the necessary UDP ports for the RTP. If you have NAT uptream in the network, you should see if in the layer 4 the IPs shown in the SIP messages got rewritten by its public IPs, it should have, or else you'll never get it working right. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended transfer to Queue
Hi, I will confess immediately that this is only tested on 1.2.24, and I would be interested to know if it happens on 1.4, but I cannot find a bug-tracker entry which represents this issue. Consider a PSTN call which comes into asterisk, and is bridged to a SIP phone. The phone operator then places the call on hold (hold music plays) and a second call is made from this handset to a Queue... Operator can now hear hold music from the queue. The operator then completes the attended transfer, bridging the initial PSTN call to the Queue. The system sees a transfer being completed, and stops MOH on both of the channels. This means that the caller is correctly transferred to a queue, but the MOH has been stopped, and they hear silence. While this behaviour is expected, it is not ideal and if anyone can point me at a workaround, or an existing bug-tracker entry, I would be most grateful. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Finding the status of an extension
Hello, I would like to check whether an extension is busy or not before calling the Dial() application to it (for example - to play a Busy if it is on conversation). How do I check it? In the trunk version there was a function DEVSTATE(SIP/123), however it does not exist on version 1.4.13... What is the equivalent of it? Thanks, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API Manager
Yes, but you should use astmanproxy instead and don't bother Asterisk with multiple manager connections. On Nov 27, 2007 8:24 AM, Anthony Chapellier [EMAIL PROTECTED] wrote: Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail Uniden UIP-200 phones
Yep, that fixed it. Just shaking my head as to why the behavior changed... Lyle CunningPike wrote: Try dtmfmode=inband CP Lyle Giese wrote: I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect' from Allison. I go to any other phone and I can log in just fine. Just not from our two Uniden phones. I have no problem placing calls. In the messages log, I see: app_voicemail.c: Unable to read password or app_voicemail.c:Couldn't read username Again, going to a different phone other than one of my two Uniden phones and no problem accessing and retreiving voicemail. In sip.conf against the UIP-200's I have: nat=never dtmfmode=rfc2833 Otherwise, I stayed with the standard Uniden provided config files served up via tftp and only made the minimum required changes to config files in Asterisk. I am running firmware 4.77(also tried downgrading firmware on phones to 4.63). Any suggestions? Thanks, Lyle Giese ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Shlomo, My understanding is I have to do a no fixup sip 5060. This from Cisco. Without doing the no fixup the registration ports get all mangled. On Nov 27, 2007 10:11 AM, Shlomo Dubrowin [EMAIL PROTECTED] wrote: Matt, If your phone is using SIP, then you should enable sip inspection (7.xcode or above) or fixup sip ( 6.x code) and have a rule that allows source (wherever you need) inbound on the outside interface to TCP 5060 (SIP port). The sip inspection or fixup should enable the proper ports for the require RTP streams. I had this working through an ASA at some point, but I don't remember if both ends were doing NAT or only one end. I don't know the phone you are talking about, but you also might want to look into STUN or ICE to get beyond the NAT Traversal issue, if that is what's causing the problem. In the Firewall log, are you seeing Denys? or drops? Have you tried debug sip on the firewall console? I've been dealing with several ASA SIP issues lately. SIP trunking with NAT will certainly not work and there is a Cisco Bug that my company discovered when setting up our PBX. Shlomo in Israel On 11/27/07, Matt [EMAIL PROTECTED] wrote: Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIXfirewall? Ports 1-2 UDP are open on the PIX and forwarding to the Asteriskserver. The Asterisk server's RTP.CONF is set to use 1-2.The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra 9133i. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
You can also create the vpn using the existing pix and netgear, eliminating more hardware and points of failure. - Original Message - From: Ricardo Carvalho [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 27, 2007 7:30:35 AM (GMT-0800) America/Los_Angeles Subject: Re: [asterisk-users] Asterisk behind a PIX firewall? Try to just open port 5060 for SIP signaling on the PIX and also enable the INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and open the necessary UDP ports for the RTP. If you have NAT uptream in the network, you should see if in the layer 4 the IPs shown in the SIP messages got rewritten by its public IPs, it should have, or else you'll never get it working right. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Hi Adrian here is what I got when I ran the command you suggested: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* I also ran netstat -l and got the following: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 *:4000 *:* LISTEN tcp0 0 *:nfs *:* LISTEN tcp0 0 *:8003 *:* LISTEN tcp0 0 *:8004 *:* LISTEN tcp0 0 *:1001 *:* LISTEN tcp0 0 *:mysql *:* LISTEN tcp0 0 *:43724 *:* LISTEN tcp0 0 *:5038 *:* LISTEN tcp0 0 *:sunrpc*:* LISTEN tcp0 0 *:sieve *:* LISTEN tcp0 0 *:http *:* LISTEN tcp0 0 *:ftp *:* LISTEN tcp0 0 *:h323hostcall *:* LISTEN tcp0 0 *:8088 *:* LISTEN tcp0 0 *:smtp *:* LISTEN tcp0 0 *:https *:* LISTEN tcp0 0 *:46302 *:* LISTEN tcp0 0 *:ssh *:* LISTEN udp0 0 *:32768 *:* udp0 0 *:nfs *:* udp0 0 *:32770 *:* udp0 0 *:2727 *:* udp0 0 *:4520 *:* udp0 0 *:5060 *:* udp0 0 *:715 *:* udp0 0 *:4569 *:* udp0 0 *:998 *:* udp0 0 *:sunrpc*:* udp0 0 192.168.1.55:ntp*:* udp0 0 localhost.localdoma:ntp *:* udp0 0 *:ntp *:* udp0 0 *:ntp *:* So does this mean that port 5060 is not listening for traffic? Thanks again, Zaheer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
On Nov 27, 2007 9:08 AM, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I disagree with you, setting in sip.conf: externhost=ddnsname;or set the next setting externip=x.x.x.x;external ip externrefresh=10;for dns localnet=192.168.0.0/255.255.0.0 should take care of this, I have never had a problem with dual nat like this, using Aastra, Cisco, Polycom and linksys. I live OpenVPN bridges for double NAT situations, of course you could try IAX2 but I have seen too many sound quality issues surrounding IAX2 so I try to stick with SIP, even if that means setting up VPNs. Thanks, Steve 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Enabling the fixup breaks the registration. On Nov 27, 2007 10:30 AM, Ricardo Carvalho [EMAIL PROTECTED] wrote: Try to just open port 5060 for SIP signaling on the PIX and also enable the INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and open the necessary UDP ports for the RTP. If you have NAT uptream in the network, you should see if in the layer 4 the IPs shown in the SIP messages got rewritten by its public IPs, it should have, or else you'll never get it working right. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding the status of an extension
Yehavi Bourvine +972-8-9489444 wrote: Hello, I would like to check whether an extension is busy or not before calling the Dial() application to it (for example - to play a Busy if it is on conversation). How do I check it? In the trunk version there was a function DEVSTATE(SIP/123), however it does not exist on version 1.4.13... What is the equivalent of it? http://www.asterisk.org/node/48360 Btw, you can use also g flag of Dial() so, that call will continue, and you will get BUSY status instantly. Regards, Atis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom phones, blinking lights and call pickup
Hi! I have the following questions/problems with * 1.4. We have several Snom phones (320 and 360). Hints are configured in extensions.conf (core show hints shows the correct values). My Snom phone is registered to some numbers (validated by using sip show subscriptions). I see the lights blinking if someone calls the subscribed number and steady lights if the call is established. So far, so good. What I want now is that I can see the number of the caller in my display and can pickup the call by pressing the blinking lights. I can pickup the call using the pickup extension. IIRC there is some pickup magic with Snom firmware 7.x. But this doesn’t solve my problem with the missing number in the display. A little example: Person A calls Person B. I am subscribed to B’s number. My Snom light is blinking. Now I want to see the number and be able to pickup the call. If I can restrict this bevaviour with pickupgroups, this would be great. When I used * 1.2, the example above would result in the displayed message „From B to B” in the phone which was quite useless (I don’t see who called), but by pressing the blinking light I could pickup the call. This is a part of my sip.conf for a snom phone: [404] type=friend context=sip callerid=”Stephan Seitz” 404 username=404 secret=secret host=dynamic defaultip=10.10.30.103 canreinvite=no mailbox=404 vmexten=1404 call-limit=1 useclientcode=yes subscribecontext = sip notifyringing = yes notifyhold=yes dtmfmode = rfc2833 Any hints how to solve the problem are welcome. Shade and sweet water! Stephan -- | Stephan SeitzE-Mail: [EMAIL PROTECTED] | | PGP Public Keys: http://fsing.rootsland.net/~stse/pgp.html | signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Restricting the manager interface to a number?
Hi! Some persons are using a TAPI driver to connect via the manager interface to the asterisk (1.4). While I can give every user his own password, I didn’t find a way to restrict a user to a certain phone number, so that he can only dial with his number via the TAPI driver and can only answer calls for another number. Is the possible? If yes, how? Thanks for the answers. Shade and sweet water! Stephan -- | Stephan SeitzE-Mail: [EMAIL PROTECTED] | | PGP Public Keys: http://fsing.rootsland.net/~stse/pgp.html | signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Hi Adrian here is what I got when I ran the command you suggested: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* I also ran netstat -l and got the following: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 *:4000 *:* LISTEN tcp0 0 *:nfs *:* LISTEN tcp0 0 *:8003 *:* LISTEN tcp0 0 *:8004 *:* LISTEN tcp0 0 *:1001 *:* LISTEN tcp0 0 *:mysql *:* LISTEN tcp0 0 *:43724 *:* LISTEN tcp0 0 *:5038 *:* LISTEN tcp0 0 *:sunrpc*:* LISTEN tcp0 0 *:sieve *:* LISTEN tcp0 0 *:http *:* LISTEN tcp0 0 *:ftp *:* LISTEN tcp0 0 *:h323hostcall *:* LISTEN tcp0 0 *:8088 *:* LISTEN tcp0 0 *:smtp *:* LISTEN tcp0 0 *:https *:* LISTEN tcp0 0 *:46302 *:* LISTEN tcp0 0 *:ssh *:* LISTEN udp0 0 *:32768 *:* udp0 0 *:nfs *:* udp0 0 *:32770 *:* udp0 0 *:2727 *:* udp0 0 *:4520 *:* udp0 0 *:5060 *:* udp0 0 *:715 *:* udp0 0 *:4569 *:* udp0 0 *:998 *:* udp0 0 *:sunrpc*:* udp0 0 192.168.1.55:ntp*:* udp0 0 localhost.localdoma:ntp *:* udp0 0 *:ntp *:* udp0 0 *:ntp *:* So does this mean that port 5060 is not listening for traffic? Thanks again, Zaheer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
Stephan Seitz wrote: We have several Snom phones (320 and 360). Hints are configured in extensions.conf (core show hints shows the correct values). My Snom phone is registered to some numbers (validated by using sip show subscriptions). I see the lights blinking if someone calls the subscribed number and steady lights if the call is established. So far, so good. What I want now is that I can see the number of the caller in my display and can pickup the call by pressing the blinking lights. I can pickup the call using the pickup extension. IIRC there is some pickup magic with Snom firmware 7.x. But this doesn’t solve my problem with the missing number in the display. Asterisk is not able to do that. I once wrote a patch for it (see bug 5014) which was not checked in. However picking up the call by pressing the button can be done with the Snom phones. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
C F wrote: On Nov 27, 2007 9:08 AM, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I disagree with you, setting in sip.conf: externhost=ddnsname;or set the next setting externip=x.x.x.x;external ip externrefresh=10;for dns localnet=192.168.0.0/255.255.0.0 should take care of this, I have never had a problem with dual nat like this, using Aastra, Cisco, Polycom and linksys. You are probably right. I think the first and last time I attempted double NATs, there was no sip.conf, I have to keep up with the times, lol. Worth a shot. I still like the OpenVPN solution for security and other added benefits. I live OpenVPN bridges for double NAT situations, of course you could try IAX2 but I have seen too many sound quality issues surrounding IAX2 so I try to stick with SIP, even if that means setting up VPNs. Thanks, Steve 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
On Tue, 27 Nov 2007, Matt wrote: Shlomo, My understanding is I have to do a no fixup sip 5060. This from Cisco. Without doing the no fixup the registration ports get all mangled. So yet another router with a broken SIP ALG... (Juniper NetScreen is one I had issues with) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Steve Totaro wrote: C F wrote: On Nov 27, 2007 9:08 AM, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I disagree with you, setting in sip.conf: externhost=ddnsname;or set the next setting externip=x.x.x.x;external ip externrefresh=10;for dns localnet=192.168.0.0/255.255.0.0 should take care of this, I have never had a problem with dual nat like this, using Aastra, Cisco, Polycom and linksys. You are probably right. I think the first and last time I attempted double NATs, there was no sip.conf, I have to keep up with the times, lol. Worth a shot. I still like the OpenVPN solution for security and other added benefits. Sorry, those options were not available in sip.conf is what I meant to say. I live OpenVPN bridges for double NAT situations, of course you could try IAX2 but I have seen too many sound quality issues surrounding IAX2 so I try to stick with SIP, even if that means setting up VPNs. Thanks, Steve 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
On 11/27/07, Stephan Seitz [EMAIL PROTECTED] wrote: [snip] So far, so good. What I want now is that I can see the number of the caller in my display and can pickup the call by pressing the blinking lights. I can pickup the call using the pickup extension. Hi, I would first ask yourself what you expect to happen if (for example) you have a snom sidecar, and there are 42 lights all blinking at the same time? I do not think that your feature request stands the usability test :) Even in the much simpler case of 2 calls coming in at once, how do you associate the numbers on the screen to the correct buttons? What if the ringing-hint is flashing for 2 or more simultaneous calls (which is quite possible) - Which one is indicated, and how do you choose which one to pick-up? Perhaps snom have come up with some fantastic method of dealing with this. Given the space available on their LCD - I personally doubt it! Just my 2p. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hostname in MySQL CDR records
On Thursday 01 November 2007 12:57:10 am Steve Edwards wrote: On Wed, 31 Oct 2007, Jim Gottlieb wrote: I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? Personally, I think the userfield is a hack. I prefer to add properly named columns to the cdrs table using cdr_addon_mysql. It makes everything so much more obvious -- especially when you don't have to cram several values into the singularly obtuse userfield. i'm sure i'm missing something. i, too, would like to add custom fields using cdr_addon_mysql, but how and where would i define them? can i add a [mappings] section to the cdr_mysql.conf file? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
On Nov 27, 2007 11:02 AM, C F [EMAIL PROTECTED] wrote: On Nov 27, 2007 9:08 AM, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I disagree with you, setting in sip.conf: externhost=ddnsname;or set the next setting externip=x.x.x.x;external ip externrefresh=10;for dns localnet=192.168.0.0/255.255.0.0 should take care of this, I have never had a problem with dual nat like this, using Aastra, Cisco, Polycom and linksys. LO! This worked! All it needed was an externip entry! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding the status of an extension
Hi,
Here is a small AGI script that get you the hint status of the extension
simply call AGI(script.agi,SIP/100)
!/usr/bin/perl
#
# page.agi - Original file was allpage.agi by Rob Thomas 2005.
# With parts of allcall.agi Original file by John Baker
# Modified by Adam Boeglin to allow for paging sccp phones
#Modified/Updated by Jeremy Betts 6/1/2006 for improved efficiency..
# We now use AGI to set the dialplan variable.. much smarter!
#Modifier by Andre Courchesne 11-27-2007 ([EMAIL PROTECTED])
# Modified to return the hint status of a single extension
# Tested with SIP extension only
#
# This program is free software; you can redistribute it and/or
# modify it under the terms of Version 2 of the GNU General
# Public License as published by the Free Software Foundation
#
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
#
#
# This works with both my aastra, polycom, sipura/linksys and cisco sccp phones.
# It should be easily modified for other sip phones
#
# Documentation:
# Simply returns channel variable MYSTATE
# 0: Extension is busy
# 1: Extension is available
#
#
#
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
@bypass = @ARGV;
@sips = `sudo /usr/sbin/asterisk -rx show hints | grep -a $bypass[0]`;
#foreach $sipline (@sips)
#{
# print Noop $sipline\n;
#}
#print Noop ---\n;
#foreach $argline (@bypass)
#{
# print Noop $argline\n;
#}
$mystate=0;
my ($junk0, $junk1, $junk2, $exten, $state, $junk2) = split(/ +/, $sips[0],6);
my ($type, $extension) = split(/\//,$chan,2);
print Noop Comparing [.$exten.] and [.$bypass[0].] $state\n;
if($state eq State:Idle)
{
$mystate=1;
print Noop Found extension $exten to be Idle\n;
}
___
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Re: [asterisk-users] Restricting the manager interface to a number?
Stephan Seitz wrote: Some persons are using a TAPI driver to connect via the manager interface to the asterisk (1.4). While I can give every user his own password, I didn’t find a way to restrict a user to a certain phone number, so that he can only dial with his number via the TAPI driver and can only answer calls for another number. Is the possible? No. Currently this type of restriction needs to be done in whatever proxy script you use to connect to the AMI. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
Steve Davies wrote: On 11/27/07, Stephan Seitz [EMAIL PROTECTED] wrote: [snip] So far, so good. What I want now is that I can see the number of the caller in my display and can pickup the call by pressing the blinking lights. I can pickup the call using the pickup extension. Hi, I would first ask yourself what you expect to happen if (for example) you have a snom sidecar, and there are 42 lights all blinking at the same time? I do not think that your feature request stands the usability test :) Even in the much simpler case of 2 calls coming in at once, how do you associate the numbers on the screen to the correct buttons? What if the ringing-hint is flashing for 2 or more simultaneous calls (which is quite possible) - Which one is indicated, and how do you choose which one to pick-up? Perhaps snom have come up with some fantastic method of dealing with this. Given the space available on their LCD - I personally doubt it! All he's asking for are SIP NOTIFYs of the extension state with dialog-info XML in the body as described in RFC 4235 and elsewhere. It's up to the SIP user agent (i.e. the phone) how to deal with it. (e.g. display only the first or last message if the screen is too small ...) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding the status of an extension
If you have some SIP phone BLF feature capable, you can try it. With it in the phone you can view the state of those the registered extensions you like, as well with it if you do sip show subscriptions in the asterisk CLI, you'll get the list of extensions and the last state of those monitored extensions. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API Manager
However I wanted to get periodic infos about queued users (position in queue) only. So I thought I could make a program sending periodic requests to asterisk manager. Is it really bad to bother asterisk manager with frequently and periodic requests sent by mutiple users (we could say maybe 100 users with a request every minute) ? Moises Silva a écrit : Yes, but you should use astmanproxy instead and don't bother Asterisk with multiple manager connections. On Nov 27, 2007 8:24 AM, Anthony Chapellier [EMAIL PROTECTED] wrote: Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
On 11/27/07, Philipp Kempgen [EMAIL PROTECTED] wrote: All he's asking for are SIP NOTIFYs of the extension state with dialog-info XML in the body as described in RFC 4235 and elsewhere. It's up to the SIP user agent (i.e. the phone) how to deal with it. (e.g. display only the first or last message if the screen is too small ...) ...and I was pointing out that in most cases, there is no point in asking for this feature, because there is often no useful way of presenting the information if you have it. :) Steve PS. The bristuff patch updates asterisk to send the extra notify information in a snom-happy format. PPS. Directed pickup on Aastra phones is broken if a full NOTIFY is sent using the bristuff patch. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt wrote: On Nov 27, 2007 11:02 AM, C F [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Nov 27, 2007 9:08 AM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Matt wrote: Just checking NAT=yes, canreinvite=no ? Correct, I have those settings set for this phone. Asterisk has been reloaded even restarted. Is this a dual NAT situation? NAT on the phone side and NAT at the PIX? If so, I fear it will never work, you might get one way audio though. I disagree with you, setting in sip.conf: externhost=ddnsname;or set the next setting externip=x.x.x.x;external ip externrefresh=10;for dns localnet=192.168.0.0/255.255.0.0 http://192.168.0.0/255.255.0.0 should take care of this, I have never had a problem with dual nat like this, using Aastra, Cisco, Polycom and linksys. LO! This worked! All it needed was an externip entry! This is good to hear. Now I know it can be done this way, although I still prefer OpenVPN for it's security and ability to let you do other things such as AMI or whatever. It is kind of hard to portscan 5060 when it is not open. I bet I could do a portscan on 5060 and of those hits try username 100 password 100 all the way up to and eventually get some toll fraud access in a day's time. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
This is good to hear. Now I know it can be done this way, although I still prefer OpenVPN for it's security and ability to let you do other things such as AMI or whatever. It is kind of hard to portscan 5060 when it is not open. I bet I could do a portscan on 5060 and of those hits try username 100 password 100 all the way up to and eventually get some toll fraud access in a day's time. GADS! I hope not! We are using fairly complex passwords :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip to ATA?
Yes, you would just plug the FXO ports directly into the cable modem and it would work. In my experience the cable media gateways (Arris and Motorola) are more robust than VoIP ATAs. They can provide polarity reversal on disconnect, can power a higher REN, and even have built-in batteries to keep the phones running when the power goes out (the battery only powers the phones, not the Internet connection). YMMV based on your local cable co's implementation. From a financial standpoint, I think you would be better off with a SIP/IAX trunk, because cable telephone service is taxed just like regular POTS service, adding 40%+ to the bill. - Original Message - From: Joe Acquisto [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, November 27, 2007 9:01 AM Subject: [asterisk-users] Sip to ATA? Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding how the phone service works. All I can get is I plug my existing phones and answering machines into the back of the cable modem and am good to go. I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
At 05:26 AM 11/27/2007, you wrote: By the way, CentOS is winning at the moment. I don't see how to publish the results live yet, but I'll do a static report as soon as there are more. Not much of a surprise, TrixBox and it's predecessors all installed CentOS by default and all the people like me for whom Asterisk was their first ever Linux install likely ended up with CentOS. No clue if it's the best, but it works, it's my phones and I've no reason to mess with it. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip to ATA?
At 06:01 AM 11/27/2007, you wrote: I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense That's what I use other than I have the 4 FXO TDM400 and it works just fine. What I got was a cable modem with 2 POTs jacks. They cut me free from the outside world at the box and plugged the modem into a wall jack and my Asterisk box. No apparent change in the working of anything. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hostname in MySQL CDR records
You can also use CDR(userfield) parameter and that way you can write in the column userfield of your CDR table of the DB, the hostname of the asterisk for each call. You can try something like the following in the dialplan of each machine: Set(CDR(userfield)=hostname_of_the_machine) Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Matt wrote: This is good to hear. Now I know it can be done this way, although I still prefer OpenVPN for it's security and ability to let you do other things such as AMI or whatever. It is kind of hard to portscan 5060 when it is not open. I bet I could do a portscan on 5060 and of those hits try username 100 password 100 all the way up to and eventually get some toll fraud access in a day's time. GADS! I hope not! We are using fairly complex passwords :) No, then you are good, but I would bet my life that there are a good many systems that use the extension for both password and username and can be accessed from the net. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Urgent question.
Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
In other words, what I need is a way for the upstream switch to somehow think that the B channels are out of service, but without actually taking the B channels out of service and dropping the existing calls. From within asterisk, zaptel, wanpipe, whatever. Is that possible? On Tue, 27 Nov 2007, Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
No, then you are good, but I would bet my life that there are a good many systems that use the extension for both password and username and can be accessed from the net. O yeah.. I can imagine.. wonder how many open systems are out there :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip to ATA?
On 11/27/2007 at 12:26 PM, Ira [EMAIL PROTECTED] wrote: At 06:01 AM 11/27/2007, you wrote: I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense Thanks for both the replies. Hope springs eternal. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind a PIX firewall?
Steve Totaro wrote: Matt wrote: This is a dual NAT situation. PIX on Asterisk side, and Netgear on phone side. HOWEVER.The Asterisk box has it's own IP but it is being tunneled through the PIX.I guess the PIX must be messing something up? If I remove the phone from behind the Netgear... then I get the audio from the Asterisk PBX so traffic seems to be flowing but why would it not get behind the firewalls? Trust me on this, I have tried almost everything to get it to work, the best you can hope for is one way audio in a dual NAT. I'm in a dual-NAT situation and it works ok... with Sipura ATAs, Linksys 941 and 841, softphones, and one polycom 330. I had to enable NAT keep alive on the Linksys/Sipuras. Ugo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? I think if you just pull the plug the PRI goes to red alarm and the provider will try one of your other PRIs for new incoming calls. Of course I don't know how your provider will handle this so don't trust me. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ResetCDR Options v, a - Asterisk 1.4
show application ResetCDR shows 3 option values which may be used. w works as expected, but what about v and a? v -- Save CDR variables - but theese are saved anyway a -- Store any stacked records - but what are stacked records? I found some ResetCDR examples using option w or no option. In which situation one of the other options does make sense? Any help appreciated -- Stefan Tichy ( asterisk at pi4tel dot de ) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
Well, right; the problem is that this also drops the existing calls in progress. :-) On Tue, 27 Nov 2007, Philipp Kempgen wrote: Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? I think if you just pull the plug the PRI goes to red alarm and the provider will try one of your other PRIs for new incoming calls. Of course I don't know how your provider will handle this so don't trust me. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
Philipp Kempgen wrote: I think if you just pull the plug the PRI goes to red alarm and the provider will try one of your other PRIs for new incoming calls. Of course I don't know how your provider will handle this so don't trust me. :) The downside of this is that established calls would be lost. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hostname in MySQL CDR records
On Tuesday 27 November 2007 11:50:09 am Ricardo Carvalho wrote: You can also use CDR(userfield) parameter and that way you can write in the column userfield of your CDR table of the DB, the hostname of the asterisk for each call. You can try something like the following in the dialplan of each machine: Set(CDR(userfield)=hostname_of_the_machine) Regards, Ricardo Carvalho. that's what i do currently, but i was wondering if it was possible to have custom cdr fields in cdr_addon_mysql as it is when using csv files. with a core show function CDR, it states: All of the above variables are read-only, except for accountcode, userfield, and amaflags. You may, however, supply a name not on the above list, and create your own variable, whose value can be changed with this function, and this variable will be stored on the cdr. but it seems like these create my own variables do not appear in mysql, though they do appear in the csv files. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
You should be able to issue a stop gracefully command to asterisk. That'll cause it to stop accepting new calls, but will let existing calls continue until complete. -erik On Nov 27, 2007 12:06 PM, Alex Balashov [EMAIL PROTECTED] wrote: In other words, what I need is a way for the upstream switch to somehow think that the B channels are out of service, but without actually taking the B channels out of service and dropping the existing calls. From within asterisk, zaptel, wanpipe, whatever. Is that possible? On Tue, 27 Nov 2007, Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Copy or Make + Make Install
Hi List; If I have a running Asterisk on one machine and I need to have another Asterisk on another machine, can I copy the files from the first running Asterisk machine to the new machine or I have to do the ./configure + make + make install? If I can copy, then which directories (and files) need to be copied? What if my new machine have other kernel version that first machine? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
Generally, best practices would dictate that you do this while the business is closed if it is not a 24/7 operation. I guess that kind of foresight comes with experience. Glad to see things are going smoothly. Thanks, Steve Erik Anderson wrote: You should be able to issue a stop gracefully command to asterisk. That'll cause it to stop accepting new calls, but will let existing calls continue until complete. -erik On Nov 27, 2007 12:06 PM, Alex Balashov [EMAIL PROTECTED] wrote: In other words, what I need is a way for the upstream switch to somehow think that the B channels are out of service, but without actually taking the B channels out of service and dropping the existing calls. From within asterisk, zaptel, wanpipe, whatever. Is that possible? On Tue, 27 Nov 2007, Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Semi-OT Part 2: Videophone
Following the recommendations here I've ordered a couple different Polycom Aastra phones to play with speaker phones. One of our next big projects is Video. I know Grandstream has a video phone, has anyone used it. Anyone have any other recommendations? Thanks, Ken (Who awaits the vmukti.com response.) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
To answer the question, there is currently no way to busy out a channel except to put it in use. There was some discussion about adding this feature at Astricon and on the list fairly recently. Thanks, Steve Steve Totaro wrote: Generally, best practices would dictate that you do this while the business is closed if it is not a 24/7 operation. I guess that kind of foresight comes with experience. Glad to see things are going smoothly. Thanks, Steve Erik Anderson wrote: You should be able to issue a stop gracefully command to asterisk. That'll cause it to stop accepting new calls, but will let existing calls continue until complete. -erik On Nov 27, 2007 12:06 PM, Alex Balashov [EMAIL PROTECTED] wrote: In other words, what I need is a way for the upstream switch to somehow think that the B channels are out of service, but without actually taking the B channels out of service and dropping the existing calls. From within asterisk, zaptel, wanpipe, whatever. Is that possible? On Tue, 27 Nov 2007, Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hostname in MySQL CDR records
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anthony Messina wrote: with a core show function CDR, it states: All of the above variables are read-only, except for accountcode, userfield, and amaflags. You may, however, supply a name not on the above list, and create your own variable, whose value can be changed with this function, and this variable will be stored on the cdr. but it seems like these create my own variables do not appear in mysql, though they do appear in the csv files. Correct. Unless you are using the new adaptive odbc driver. I actually use a patch in house for this, but its definitely a hack and I normally just apply it manually. Obviously it wasn't accepted into SVN, but we try to submit everything we work on: http://bugs.digium.com/view.php?id=9424 - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTGrLDQNt8rg0Kp4RAqsUAJwLC0VfAi4a8ilrB+K0t564+r6GgACeI5o8 AUlI25dlKmJsNid6MQ42TNc= =4s6G -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 randulo wrote: NEW Asterisk favorite OS survey: http://www.esurveyspro.com/Survey.aspx?id=1f38482e-c3fa-4384-8b8a-65a9f40b2cd8 The above is a compromise between three elements: 1) The amount of time and patience I have 2) The information I think most people are willing to proovide 3) Steve's requests Please have at it. Steve, you can use this same setup free to design your own. Where are the results? Once I posted it just redirected to http://food4wine.ning.com/ - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTGw9DQNt8rg0Kp4RAmebAJ933Wmc9zIrZhoGUPB/cUwakYAQfgCdHpno a94QpZAo5kFGlJ8txVeiBIE= =oZmU -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API Manager
So you'd be making 100 connections/minute, which is pretty relentless. That's like five connections, five requests sent, five responses received, and five disconnects, /every/ three seconds. And the likelihood of all 100 users to be spread out evenly over a minute doesn't seem very high. I think your box would be pretty busy with that. Astmanproxy would be indicated. Moj Anthony Chapellier wrote: However I wanted to get periodic infos about queued users (position in queue) only. So I thought I could make a program sending periodic requests to asterisk manager. Is it really bad to bother asterisk manager with frequently and periodic requests sent by mutiple users (we could say maybe 100 users with a request every minute) ? Moises Silva a écrit : Yes, but you should use astmanproxy instead and don't bother Asterisk with multiple manager connections. On Nov 27, 2007 8:24 AM, Anthony Chapellier [EMAIL PROTECTED] wrote: Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT Part 2: Videophone
We've used the Grandstream video phone quite a bit, and I have to say, I'm considerably impressed with its quality. YES, it's a Grandstream (and has the usual quirks and annoyances that one has come to expect now and again), but the quality of the screen and camera are both excellent, and with the later firmware handling H263 as well as H264, it meshes quite well with many of the software video-capable phones out there such as Ekiga and X-Lite v3/eyeBeam. Sorry... no vmukti.com response from me, I'm afraid. N. Ken Williams wrote: Following the recommendations here I've ordered a couple different Polycom Aastra phones to play with speaker phones. One of our next big projects is Video. I know Grandstream has a video phone, has anyone used it. Anyone have any other recommendations? Thanks, Ken (Who awaits the vmukti.com response.) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
Results should be withheld until the end of the specified polling period. Thanks, Steve Totaro 888.777.1888 Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 randulo wrote: NEW Asterisk favorite OS survey: http://www.esurveyspro.com/Survey.aspx?id=1f38482e-c3fa-4384-8b8a-65a9f40b2cd8 The above is a compromise between three elements: 1) The amount of time and patience I have 2) The information I think most people are willing to proovide 3) Steve's requests Please have at it. Steve, you can use this same setup free to design your own. Where are the results? Once I posted it just redirected to http://food4wine.ning.com/ - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTGw9DQNt8rg0Kp4RAmebAJ933Wmc9zIrZhoGUPB/cUwakYAQfgCdHpno a94QpZAo5kFGlJ8txVeiBIE= =oZmU -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
On Nov 27, 2007 8:13 PM, Matt Riddell [EMAIL PROTECTED] wrote: Where are the results? Hey Matt, As I said above, this site doesn't appear to allow public results page. (Actually I said I couldn't find one) I'll show the static one as soon as I can. Wait, this is the Internet. I can go now. Ok, I'm back. This software kinda sucks (great value for the price of zero, but that's it.) There is no way to see the data except as a CSV. Sorry this is lame, but here's the file. I'll update it tomorrow (it's night here ,now). http://voipusersconference.org/poll/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.2.22 and 1.4.7 released
The Asterisk.org development team has announced the release of Zaptel versions 1.2.22 and 1.4.7. These releases contain (among other things) many bug fixes to the TC400B driver, a bug fix on the wctdm24xxp driver for users with a VPM150M, as well as numerous improvements and fixes to the Xorcom driver suite. The much better performing version of fxotune from 1.4 has now been put into 1.2, so you may wish to rerun this tool with the new version. As always, please see the respective Changelogs for additional information. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Thank you for your support! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
randulo wrote: On Nov 27, 2007 8:13 PM, Matt Riddell [EMAIL PROTECTED] wrote: Where are the results? Hey Matt, As I said above, this site doesn't appear to allow public results page. (Actually I said I couldn't find one) I'll show the static one as soon as I can. Wait, this is the Internet. I can go now. Ok, I'm back. This software kinda sucks (great value for the price of zero, but that's it.) There is no way to see the data except as a CSV. Sorry this is lame, but here's the file. I'll update it tomorrow (it's night here ,now). http://voipusersconference.org/poll/ not ready yet! LOL. I like your style Randulo ;-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Copy or Make + Make Install
Hello, Only copy the configuration files, extensions.conf, sip.conf, iax.conf , Best regards On Nov 27, 2007 1:27 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I have a running Asterisk on one machine and I need to have another Asterisk on another machine, can I copy the files from the first running Asterisk machine to the new machine or I have to do the ./configure + make + make install? If I can copy, then which directories (and files) need to be copied? What if my new machine have other kernel version that first machine? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
On Nov 27, 2007 8:17 PM, Steve Totaro [EMAIL PROTECTED] wrote: Results should be withheld until the end of the specified polling period. Gh! Ok, I'll take 'em off. We can give this a few days then , and when things stop, I'll publish. Open to suggestions. As far as the open source project mentioned earlier, you have to register for it and I don't feel like installing something like this on a server since it hasn't been touched in two years and is not useful to me or myy customers. Wufoo, the original software used, is really excellent. Too bad they insist on being paid for such good work :) I will use them for a paying customer who needs this service. I'll report the findings this Friday on the conference as well, obviously. http://voipusersconference.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
So can anyone tell me why the * server is not listening on UDP port 5060? Thanks, Zaheer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: Tuesday, November 27, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP port 5060 closed - how do I open it? Hi Adrian here is what I got when I ran the command you suggested: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* I also ran netstat -l and got the following: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 *:4000 *:* LISTEN tcp0 0 *:nfs *:* LISTEN tcp0 0 *:8003 *:* LISTEN tcp0 0 *:8004 *:* LISTEN tcp0 0 *:1001 *:* LISTEN tcp0 0 *:mysql *:* LISTEN tcp0 0 *:43724 *:* LISTEN tcp0 0 *:5038 *:* LISTEN tcp0 0 *:sunrpc*:* LISTEN tcp0 0 *:sieve *:* LISTEN tcp0 0 *:http *:* LISTEN tcp0 0 *:ftp *:* LISTEN tcp0 0 *:h323hostcall *:* LISTEN tcp0 0 *:8088 *:* LISTEN tcp0 0 *:smtp *:* LISTEN tcp0 0 *:https *:* LISTEN tcp0 0 *:46302 *:* LISTEN tcp0 0 *:ssh *:* LISTEN udp0 0 *:32768 *:* udp0 0 *:nfs *:* udp0 0 *:32770 *:* udp0 0 *:2727 *:* udp0 0 *:4520 *:* udp0 0 *:5060 *:* udp0 0 *:715 *:* udp0 0 *:4569 *:* udp0 0 *:998 *:* udp0 0 *:sunrpc*:* udp0 0 192.168.1.55:ntp*:* udp0 0 localhost.localdoma:ntp *:* udp0 0 *:ntp *:* udp0 0 *:ntp *:* So does this mean that port 5060 is not listening for traffic? Thanks again, Zaheer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Does a sip.conf exist? Asterisk won't start the SIP UAS if it is missing. On Tue, 27 Nov 2007, Zaheer K. Master wrote: So can anyone tell me why the * server is not listening on UDP port 5060? Thanks, Zaheer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K. Master Sent: Tuesday, November 27, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP port 5060 closed - how do I open it? Hi Adrian here is what I got when I ran the command you suggested: netstat -an|grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* I also ran netstat -l and got the following: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 *:4000 *:* LISTEN tcp0 0 *:nfs *:* LISTEN tcp0 0 *:8003 *:* LISTEN tcp0 0 *:8004 *:* LISTEN tcp0 0 *:1001 *:* LISTEN tcp0 0 *:mysql *:* LISTEN tcp0 0 *:43724 *:* LISTEN tcp0 0 *:5038 *:* LISTEN tcp0 0 *:sunrpc*:* LISTEN tcp0 0 *:sieve *:* LISTEN tcp0 0 *:http *:* LISTEN tcp0 0 *:ftp *:* LISTEN tcp0 0 *:h323hostcall *:* LISTEN tcp0 0 *:8088 *:* LISTEN tcp0 0 *:smtp *:* LISTEN tcp0 0 *:https *:* LISTEN tcp0 0 *:46302 *:* LISTEN tcp0 0 *:ssh *:* LISTEN udp0 0 *:32768 *:* udp0 0 *:nfs *:* udp0 0 *:32770 *:* udp0 0 *:2727 *:* udp0 0 *:4520 *:* udp0 0 *:5060 *:* udp0 0 *:715 *:* udp0 0 *:4569 *:* udp0 0 *:998 *:* udp0 0 *:sunrpc*:* udp0 0 192.168.1.55:ntp*:* udp0 0 localhost.localdoma:ntp *:* udp0 0 *:ntp *:* udp0 0 *:ntp *:* So does this mean that port 5060 is not listening for traffic? Thanks again, Zaheer -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version survey
randulo wrote: On Nov 27, 2007 8:17 PM, Steve Totaro [EMAIL PROTECTED] wrote: Results should be withheld until the end of the specified polling period. Gh! Ok, I'll take 'em off. We can give this a few days then , and when things stop, I'll publish. Open to suggestions. As far as the open source project mentioned earlier, you have to register for it and I don't feel like installing something like this on a server since it hasn't been touched in two years and is not useful to me or myy customers. Wufoo, the original software used, is http://www.limesurvey.org/ Randy, There must be something in the water (or wine) in France. Nothing on the limesurvey site requires you to register for anything. It is very current and updated about once a month (far from abandoned). Perhaps someone else made a different suggestions. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
On Tue, Nov 27, 2007 at 04:36:33PM +, Steve Davies wrote: I would first ask yourself what you expect to happen if (for example) you have a snom sidecar, and there are 42 lights all blinking at the same time? I do not think that your feature request stands the usability test :) Well, of course if you want to monitor all lines in a call center, you will certainly have some problems. But in small environments this is working. The ISDN PBX in our company building does the following: - some phones in the office area are monitor phones for a range of numbers (about 5); - now one of these numbers are called: 1. the phone with this number rings; 2. if the call is not answered after a short time, a light (phone layout is similiar to the Snom phones) begins to blink at the monitor phones; caller ID and the target number are displayed; if you press the button next to the blinking light, you pick-up the call; 3. still a little time later the monitor phones are additionally ringing, but you still have to press the button to pick-up the call; - and while others without monitor phones may see that a line is ringing/busy, they are not allowed to take the call; Any plans of getting rid of the building PBX shared by other companys as well and replacing it with Asterix (a wish to gain more independance) are not allowed in presenting fewer features. Those are the guidelines, now I’m testing. If I can’t implement the old behaviour close enough, I will probably have to look at other solutions. Shade and sweet water! Stephan -- | Stephan SeitzE-Mail: [EMAIL PROTECTED] | | PGP Public Keys: http://fsing.rootsland.net/~stse/pgp.html | signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Yes I have a sip.conf, contents as follows: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allowexternaldomains=no allowexternalinvites=no allowguest=no allowsubscribe=no allowtransfer=yes alwaysauthreject=no autodomain=no callevents=no canreinvite=nonat compactheaders=no dumphistory=no externip=75.127.218.82 g726nonstandard=no ignoreregexpire=no jbenable=no jbforce=no jblog=no localnet=192.168.0.0/255.255.0.0 maxcallbitrate=384 maxexpiry=3600 minexpiry=60 nat=yes notifyringing=no pedantic=no promiscredir=no recordhistory=yes relaxdtmf=no rtcachefriends=no rtsavesysname=no rtupdate=no progressinband=no sendrpid=no sipdebug=yes t1min=100 t38pt_udptl=no trustrpid=no usereqphone=no videosupport=no allow=ulaw disallow=all [authentication] -Original Message- Does a sip.conf exist? Asterisk won't start the SIP UAS if it is missing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restricting the manager interface to a number?
On Tue, Nov 27, 2007 at 05:22:59PM +0100, Philipp Kempgen wrote: Currently this type of restriction needs to be done in whatever proxy script you use to connect to the AMI. Okay, thanks for the answer. Shade and sweet water! Stephan -- | Stephan SeitzE-Mail: [EMAIL PROTECTED] | | PGP Public Keys: http://fsing.rootsland.net/~stse/pgp.html | signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
Stephan Seitz wrote: The ISDN PBX in our company building does the following: - some phones in the office area are monitor phones for a range of numbers (about 5); In Asterisk's terms that is a pickup group. - now one of these numbers are called: 1. the phone with this number rings; 2. if the call is not answered after a short time, a light (phone layout is similiar to the Snom phones) begins to blink at the monitor phones; caller ID and the target number are displayed; if you press the button next to the blinking light, you pick-up the call; You could use one of the Snom pickup patches to have the callerid transmitted. (Although as an Asterisk consultant I would not recommend to use a patch.) But there is probably no solution to the delay. The light on the Snom would start to blink instantly. Not sure if this comes close enough to the behavior of your old PBX. 3. still a little time later the monitor phones are additionally ringing, but you still have to press the button to pick-up the call; - and while others without monitor phones may see that a line is ringing/busy, they are not allowed to take the call; Maybe you could dial the extension for 60 seconds or something and if they did not answer the phone put the call into a queue with the r parameter (ring instead of MOH). Any plans of getting rid of the building PBX shared by other companys as well and replacing it with Asterix (a wish to gain more independance) are not allowed in presenting fewer features. Those are the guidelines, now I’m testing. If I can’t implement the old behaviour close enough, I will probably have to look at other solutions. Grüße, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lost setting up IAXmodem after drive crash
A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email procmail faxmail iaxmodem asterisk sipext ATA (with attached fax). I have followed all the instructions on creating the IAX extension and configuring the IAXmodem config file. I can see the connection in Asterisk: iax2 show peers Name/UsernameHost Mask Port Status 24729127.0.0.1 (D) 255.255.255.255 32770 Unmonitored 1 iax2 peers [0 online, 0 offline, 1 unmonitored] at this point I am SUPPOSE to be able to test with sendfax: sendfax -d 2201 /etc/hylafax/hyla.conf and nothing happens that I can tell. I see some files in spool/hylafax: ls docq/ -lstr total 24 4 -rw--- 1 uucp 60002 1 Nov 27 15:04 seqf 8 -rw-r- 1 uucp 60002 6636 Nov 27 15:04 doc1.ps 12 -rw-r- 1 uucp 60002 11264 Nov 27 15:04 cover1 Then I found in tmp: cat tmp/ttyIAX0_last_wedged_email 1196204456 This is a temp file written and read by bin/wedged to rate-limit the emails it sends about the wedged status of device /dev/ttyIAX0. The first line contains a timestamp for when the last email was sent. This file is never deleted automatically, there's no need to do it and it may be useful to know when/if a device had last a problem. However, you can safely delete it at any time if you wish, it will be recreated when needed. last modified: Tue Nov 27 18:00:56 EST 2007 What have I left out? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
Am Montag, den 26.11.2007, 22:39 -0500 schrieb [EMAIL PROTECTED]: On Nov 26, 2007 9:52 AM, Alberto Pastore [EMAIL PROTECTED] wrote: I also found the Pirelli DP-L10 dual phone to be an excellent sip client with good roaming support and discrete battery saving capability. (Used in a 14-cell wifi network with 40 cellphones). I don't know what to say I have not used the Pirelli phone but at the same time it is the same ODM as most of the Linksys and D-Link phone and I have not been too pleased with those. They work. They roam ok but they also lock up every so often and the call quality isnt the best. You can tell the G729 codec is very taxing on the device it can take 2 sec for the phone to respond to a keypress. Hello *, I have the Pirelli phone (there are two actually, I have the bar-type one, I think it's L10) in daily use, both GSM (O2 Germany) and WLAN (registered to Asterisk of course - behind OpenWRT boxes, FritzBoxes, D-Link APs whatever is there). I had the latest firmware in August, did not check back since. In my opinion, this phone is not ready for production use for regular users. It works really nice in the short run, but a few things make it unacceptable or at least lack for my approval as a well-done product: - Connection loss on DHCP expiry (twice yet) - Relatively poor Wifi signal strength, compared to other Wifi devices - frequent lockups, which require battery removal and clock reprogramming: - if you power on the phone while it is on charger power - if you receive a WIFI call while you have a WLAN call, and the WIFI call is from the same contact in the phone book - plugging in, unplugging, plugging in headset fast in a row - no SIP voicemail support, GSM voicebox only (pressing 1) - slow user interface It further lacks - one-touch silent mode - you can only kind of emulate that - proper headset support - any key accept call does _not_ work, although it is a separate choice from accept key accept call. Auto-accept works OK though. Vibra seems to _not_ work once a headset is plugged in... or at least not always. - one-digit press in main menu to open the menu: It works in the submenues, but you always need to navigate the main menu with the arrow controls - quick mode switch WLAN on/off - if you are out of home range, WLAN seems to suck lots of battery. - display of CALLERID(name), currently only CALLERID(num) is displayed I would also like a modus which is if no known network is in range, connect to any unencrypted network you can get that seems to have network connectivity. This should of course be optional. That said, for my personal uses it is OK, lightyears in front of the two UTStarcom phones I also had in daily use. Well, while they worked anyway. Both the good WPA support (basically broken in the UTS) and the GSM function make me like it. The phone book (multiple entries) is great, although it would be nice to see from which of the numbers listed the call is coming (call Sam back on his mobile, or is he at home?) I believe most of the problems I see could be solved in software quite easily, but until they are, I would not give it to my users, rather I would go with DECT, Siemens Gigaset ISDN, on FritzBoxen internal S0 bus, because this combination I know works absolutely perfect, as good as ISDN, and that means a lot. I have been using the Pirelli for five or six months, and keep using it because the GSM/WLAN combo is just the killer app on it. It sucks a bit more than my regular mobile phone (which I sometimes carry instead, I have a Dual-SIM contract), but for me all mobiles suck. I have thin fingers, but using mobile keypads always makes me feel like having jelly sticks on my hands. That is why I love the BudgeTone 100 phone :-P Best regards, Anselm P.S.: Your fingers are too fat to dial - please mash the keys for your free dialing wand -- phone announcement in Simpsons King Size Homer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman: Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Hi Robert, I researched for something similar about a year ago, and came up with nothing really worth the work. If you can, try to get another ATA that has a real, old-fashioned serial modem plugged into it, and limit that modem to 9600. I think more than that will not work reliably, but you could of course try. The only working implementation of software emulating a modem in conjunction with asterisk I have seen is fax-related, and even there I read from several people that anything better than 9600 is hardly ever achieved. The code there is cranked into fax-use though, not modem use, which would require the PPP bytestream to be off-handed instead of fax parsing. Perhaps iaxmodem would do that No idea. I'd be interested in how you get that working, if you do indeed. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Zaheer, On 28/11/07 9:28 AM, Zaheer K. Master [EMAIL PROTECTED] wrote: Yes I have a sip.conf, contents as follows: From the CLI can you confirm SIP is running by pasting the results of 'module show like sip' Cheers Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
Hi! 2. if the call is not answered after a short time, a light (phone layout is similiar to the Snom phones) begins to blink at the monitor phones; caller ID and the target number are displayed; if you press the button next to the blinking light, you pick-up the call; I don't think you will be able to get the exact same behaviour arranged with asterisk and snom, however there are three choices for you that get quite close: 1. Use group dial like in Dial(SIP/1SIP/2) and have your monitor phones each act as SIP/2 to SIP/6 with dedicated (!) lines that have their ringer set to silent. You might want to adjust the Caller ID name to prefix it with the called number like to 123: from 4567890. The SNOMs have 12 lines, so why not actually use some of them ... A variation of this: Record a new ringer sound that is a) long and b) has 10+ seconds silence at the beginning, and let the monitoring SNOMs use that. 2. Take a closer look at the new SLA functionality in Asterisk 1.4 (shared line appearance), the Wiki has it all including a link to the blog that explains it all in detail 3. Stay with your current solution, and add a SIP MESSAGE sent using sipsak using System() that informs the monitoring phones of the caller ID; I haven't tested this particular case so I am not sure if this SIP MESSAGE would not be immediately overwritten on the phone's display with other data like (useless) pick-up information. Cheers, Philipp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lost setting up IAXmodem after drive crash
oops. I meant to post this to the Hylafax list :-[ And with a reboot, at least it made the call. Now to get the fax to print! Robert Moskowitz wrote: A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email procmail faxmail iaxmodem asterisk sipext ATA (with attached fax). I have followed all the instructions on creating the IAX extension and configuring the IAXmodem config file. I can see the connection in Asterisk: iax2 show peers Name/UsernameHost Mask Port Status 24729127.0.0.1 (D) 255.255.255.255 32770 Unmonitored 1 iax2 peers [0 online, 0 offline, 1 unmonitored] at this point I am SUPPOSE to be able to test with sendfax: sendfax -d 2201 /etc/hylafax/hyla.conf and nothing happens that I can tell. I see some files in spool/hylafax: ls docq/ -lstr total 24 4 -rw--- 1 uucp 60002 1 Nov 27 15:04 seqf 8 -rw-r- 1 uucp 60002 6636 Nov 27 15:04 doc1.ps 12 -rw-r- 1 uucp 60002 11264 Nov 27 15:04 cover1 Then I found in tmp: cat tmp/ttyIAX0_last_wedged_email 1196204456 This is a temp file written and read by bin/wedged to rate-limit the emails it sends about the wedged status of device /dev/ttyIAX0. The first line contains a timestamp for when the last email was sent. This file is never deleted automatically, there's no need to do it and it may be useful to know when/if a device had last a problem. However, you can safely delete it at any time if you wish, it will be recreated when needed. last modified: Tue Nov 27 18:00:56 EST 2007 What have I left out? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
If the hunt-group is properly done, you should be able to busy-out members of a trunk for maintenance. Otherwise, if the individual trunks have numbers (unpublished) assigned to all the circuits in the group, you could always send a Redirect() to that any of the other trunks' numbers. -Philip Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Urgent question.
Anyone have an application to robo-dial an outgoing conference call? ;-) You could tie up all your circuits with outbound calls... If you hairpin them at the switch, you shouldn't incur any usage costs... Steve Totaro wrote: To answer the question, there is currently no way to busy out a channel except to put it in use. There was some discussion about adding this feature at Astricon and on the list fairly recently. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones, blinking lights and call pickup
Philipp von Klitzing wrote: 2. if the call is not answered after a short time, a light (phone layout is similiar to the Snom phones) begins to blink at the monitor phones; caller ID and the target number are displayed; if you press the button next to the blinking light, you pick-up the call; I don't think you will be able to get the exact same behaviour arranged with asterisk and snom, however there are three choices for you that get quite close: 1. Use group dial like in Dial(SIP/1SIP/2) and have your monitor phones each act as SIP/2 to SIP/6 with dedicated (!) lines that have their ringer set to silent. You might want to adjust the Caller ID name to prefix it with the called number like to 123: from 4567890. The SNOMs have 12 lines, so why not actually use some of them ... I wouldn't like that personally but it's an interesting idea anyway. A variation of this: Record a new ringer sound that is a) long and b) has 10+ seconds silence at the beginning, and let the monitoring SNOMs use that. Might work if the sound file is on the phone before you try to dial to it. If you use the Alert-Info header in combination with ringtones longer than just a few seconds the Snom crashes. Better use the ring_after_delay setting if you plan to go this way. http://wiki.snom.com/Web_Interface/Settings/Common#ring_after_delay 3. Stay with your current solution, and add a SIP MESSAGE sent using sipsak using System() that informs the monitoring phones of the caller ID; I haven't tested this particular case so I am not sure if this SIP MESSAGE would not be immediately overwritten on the phone's display with other data like (useless) pick-up information. Might work but it smells like an ugly hack. Remember you need to clear the desktop messages. Grüße, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
