[asterisk-users] Link2VoIP going out of business! Now what?
Last week I got an email from Link2VoIP saying that they are shutting it down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. I use Link2VoIP for termination, connecting my Asterisk servers to the regular old telephone company. I like Link2VoIP, I have a few numbers with them and many of my clients do to. Anyone else being affected by this? What are you doing for VoIP termination? I am in Canada, many popular VoIP providers do not work here. And soon that number will be one less. -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Congestion outbound only with ATA boxes
I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call out. It could not call out to the Link2VoIP or any of the SIP phones. I spent a lot of time going over the configureation for this Asterisk server and the settings in the Linksys PAP2T box but could not get it to work. I removed the Linksys PAP2T and replaced it with an HT503 because I read a lot of good recommendations for this device. It seems to have almost the same problem. I say almost because when the Linksys would get congestion I would hear the Asterisk recording tell me All circuits are busy now, good-bye but the HT503 only gets a busy tone. All the SIP phones can call out no problem but these two ATA boxes that I am trying to use the FXS ports to connect old analog POTS phones to are not working. I have turned on the debug in Asterisk and can see the point where I get congestion but I don't know how to make Asterisk give me more details as to why I am getting congestion. Can anyone help me to get more details about this problem? I traced the debug from a working SIP phone as it makes an outgoing call and from the HT503 as it tries to make a call. Everything is identical right up to the point where the HT503 gets a congestion instruction from the Asterisk server. Here is the debug output just at the point where it happens. -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial(SIP/302-08221a38, SIP/301||tr) in new stack -- Called 301 Home*CLI --- Transmitting (NAT) to 192.168.0.100:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.100:5060 ;branch=z9hG4bK443855200;received=192.168.0.100;rport=5060 From: sip:302@192.168.0.1;tag=1257222779 To: sip:301@192.168.0.1;tag=as201c8013 Call-ID: 979693319-5060-5@192.168.0.100 CSeq: 41 INVITE User-Agent: FPBX-2.4.0(1.4.19.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:301@192.168.0.1 Content-Length: 0 -- SIP/301-0822de30 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dial:8] Set(SIP/302-08221a38, DIALSTATUS=CONGESTION) in new stack -- Executing [s@macro-exten-vm:10] Set(SIP/302-08221a38, SV_DIALSTATUS=CONGESTION) in new stack -- Executing [s@macro-exten-vm:11] GosubIf(SIP/302-08221a38, 0?docfu|1) in new stack -- Executing [s@macro-exten-vm:12] GosubIf(SIP/302-08221a38, 0?docfb|1) in new stack -- Executing [s@macro-exten-vm:13] Set(SIP/302-08221a38, DIALSTATUS=CONGESTION) in new stack -- Executing [s@macro-exten-vm:14] NoOp(SIP/302-08221a38, Voicemail is novm) in new stack -- Executing [s@macro-exten-vm:15] GotoIf(SIP/302-08221a38, 1?s-CONGESTION|1) in new stack -- Goto (macro-exten-vm,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-exten-vm:1] PlayTones(SIP/302-08221a38, congestion) in new stack Audio is at 192.168.0.1 port 10162 Adding codec 0x100 (g729) to SDP -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.
IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro [EMAIL PROTECTED] wrote: Check for IRQ issues, move the card to a different slot. You could ask permission to record calls so maybe you can hear the sound yourself. I would then go ahead and swap out cards. I have had TDM400 with bad modules and also bad ports on the cards themselves, so it could a hardware issue. This is what I suspect, especially if you did not put any surge suppression on your telco lines. Usually, at least in my experience, ticks or beeps indicate IRQ, hissing or loud static indicate something with/on the board is bad. ALWAYS use surge suppression on your lines! Thanks, Steve Totaro On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther [EMAIL PROTECTED] wrote: I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are there known causes for this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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 -- http://www.Radados.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] Customer complains of noise on line I cannot reproduce.
[EMAIL PROTECTED] lspci -v -s 01:07.0 pcilib: Cannot open /sys/bus/pci/devices :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ac00 [size=256] Memory at fbfff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 This is all I know about it. The client bought them about a month ago and installed them himself then asked me to setup the Asterisk program for him. The problem motherboard was a four year old Gigabyte with a Promise IDE controller. The new motherboard that works well is an ASUS but I don't know anything else about it. On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro [EMAIL PROTECTED] wrote: Which revision of the Digium TDM400? On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther [EMAIL PROTECTED] wrote: IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro [EMAIL PROTECTED] wrote: Check for IRQ issues, move the card to a different slot. You could ask permission to record calls so maybe you can hear the sound yourself. I would then go ahead and swap out cards. I have had TDM400 with bad modules and also bad ports on the cards themselves, so it could a hardware issue. This is what I suspect, especially if you did not put any surge suppression on your telco lines. Usually, at least in my experience, ticks or beeps indicate IRQ, hissing or loud static indicate something with/on the board is bad. ALWAYS use surge suppression on your lines! Thanks, Steve Totaro On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther [EMAIL PROTECTED] wrote: I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are there known causes for this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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 -- http://www.Radados.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 ___ -- 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 -- http://www.Radados.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] Customer complains of noise on line I cannot reproduce.
Not sure what you are asking for here? It looks like you have a config file that sets a clock speed but I do not know what file that is. How can I find this informaiton? On Sat, Mar 8, 2008 at 5:48 PM, Fons van der Beek [EMAIL PROTECTED] wrote: what clock? rxclock crystalclock I currently use card=1,0x4 -- http://www.Radados.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] Customer complains of noise on line I cannot reproduce.
[EMAIL PROTECTED] cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 FXSKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) Span 2: WCTDM/1 Wildcard TDM400P REV I Board 2 5 WCTDM/1/0 FXOKS (In use) 6 WCTDM/1/1 FXSKS (In use) 7 WCTDM/1/2 FXSKS (In use) 8 WCTDM/1/3 FXSKS (In use) [EMAIL PROTECTED] On Sat, Mar 8, 2008 at 7:08 PM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: paste output of head -n 1 /proc/zaptel/* 2008/3/9, Fons van der Beek [EMAIL PROTECTED]: what clock? rxclock crystalclock I currently use card=1,0x4 Grygoriy Dobrovolskyy schreef: Well i have installed asterisk on spare system to replace old one, with new tyan motherboard, surprise came when i installed digium fxs/fxo and b410 p, system unstable, random bips on start, misdn module Not loading, heh, old system worked on asus p5nd2 sli without a problem with them. There were 2 reasons why i wanted change mobo 1: chipset on asus card generated too much heat, 2: i had a new 3ware controller. So you never know really. Loud scrathing sound? sometimes a card problem, try on other hardware. Pci interrupts, also maybe sync problem (you can enable b410 clock in misdn-init.conf) Also turn off all sound/usb/etc unused devices. 2008/3/8, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] : [EMAIL PROTECTED] lspci -v -s 01:07.0 pcilib: Cannot open /sys/bus/pci/devices :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ac00 [size=256] Memory at fbfff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 This is all I know about it. The client bought them about a month ago and installed them himself then asked me to setup the Asterisk program for him. The problem motherboard was a four year old Gigabyte with a Promise IDE controller. The new motherboard that works well is an ASUS but I don't know anything else about it. On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which revision of the Digium TDM400? On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: IRQ's seem to have been the problem. Thanks Steve Totaro for that tip. The Digium cards were at the same IRQ as the IDE controller, I moved the cards and hard drives to a different system and all is good now. Thanks. On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Check for IRQ issues, move the card to a different slot. You could ask permission to record calls so maybe you can hear the sound yourself. I would then go ahead and swap out cards. I have had TDM400 with bad modules and also bad ports on the cards themselves, so it could a hardware issue. This is what I suspect, especially if you did not put any surge suppression on your telco lines. Usually, at least in my experience, ticks or beeps indicate IRQ, hissing or loud static indicate something with/on the board is bad. ALWAYS use surge suppression on your lines! Thanks, Steve Totaro On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are there known causes for this? -- Open Source: To innovate then create Proprietary: To imitate then litigate
[asterisk-users] Customer complains of noise on line I cannot reproduce.
I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are there known causes for this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] IAX2 trunks unreliable becoming UNREACHABLE after a time
I may have found a solution to why this problem is happening to me. All my IAX trunks are up and working and have been for over a day now. If there are still up and running with no problems in a week I will post again and let everyone know. At this point in time it seems the problem was caused by a poorly constructed initrd file. My servers all run RAID-1 and my /var/ is mounted on it's own RAID-1. I hand crafted a new initrd to ensure that RAID-1 started properly and that /var/ was mounted before init runs, it was not before. As of now I am very hopeful that the problem is gone, all indicators are that this was the solution I needed. On Sun, Feb 10, 2008 at 12:33 PM, Royce Souther [EMAIL PROTECTED] wrote: I have a network of offices using Asterisk that are connected via IAX2 trunks. The trunks work great for a day or two then for no reason at all one end of the trunk will become UNREACHABLE while the other end is still connected. The only way to fix the problem is to shutdown Asterisk completly then start it backup again. The end that dies is not always the same, some times it is server A and some times it is server B. Never have I seen that both ends die, just one. The side that is still connected can make calls to the end that died but not the other way. If you call from the server with the dead IAX2 trunk you here All circuts are busy now. All networks have static IP addresses and their firewalls are setup to allow UDP 4569 to come in to the Asterisk systems. I have been doing a lot of research into this problem. I found this bug tracker http://bugs.digium.com/view.php?id=5912 that talks about it being an old problem with version 1.2.1 using rand() and it not being thread safe. This I can understand. The thread proposed using rand_r() or ast_random() in place of rand(), that sounds like a good idea. So when I look at my newer 1.2.18 version I find that it is still using rand() and the bug tracker continues to be opened and closed and reopened again and again. Do I dare ask if anyone has a reliable IAX2 trunk? If so how? Should I avoid using IAX2 all together? I know SIP trunking is an option but it becomes a real management problem with trying to deal with all the many ports that need to be open through the firewalls, IAX2 seems like a better way to go if only it was reliable. -- Open Source: To innovate then create Proprietary: To imitate then litigate -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] Southern Alberta Canada * users.
Are you in the Southern Alberta area? I am putting on a free VoIP * workshop on Friday afternoon. Everyone is welcome to attend. This is to introduce local business to the benefits of VoIP using Asterisk. If you want to attend or if you have clients you think could benefit from this please email me directly for more information. -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] IAX2 trunks unreliable becoming UNREACHABLE after a time
I have a network of offices using Asterisk that are connected via IAX2 trunks. The trunks work great for a day or two then for no reason at all one end of the trunk will become UNREACHABLE while the other end is still connected. The only way to fix the problem is to shutdown Asterisk completly then start it backup again. The end that dies is not always the same, some times it is server A and some times it is server B. Never have I seen that both ends die, just one. The side that is still connected can make calls to the end that died but not the other way. If you call from the server with the dead IAX2 trunk you here All circuts are busy now. All networks have static IP addresses and their firewalls are setup to allow UDP 4569 to come in to the Asterisk systems. I have been doing a lot of research into this problem. I found this bug tracker http://bugs.digium.com/view.php?id=5912 that talks about it being an old problem with version 1.2.1 using rand() and it not being thread safe. This I can understand. The thread proposed using rand_r() or ast_random() in place of rand(), that sounds like a good idea. So when I look at my newer 1.2.18 version I find that it is still using rand() and the bug tracker continues to be opened and closed and reopened again and again. Do I dare ask if anyone has a reliable IAX2 trunk? If so how? Should I avoid using IAX2 all together? I know SIP trunking is an option but it becomes a real management problem with trying to deal with all the many ports that need to be open through the firewalls, IAX2 seems like a better way to go if only it was reliable. -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] Need a dial rule to match and replace a number.
I am using Asterisk 1.2.18 with FreePBX 2.2.0. I have two Asterisk systems with an IAX2 trunk between them. I want to make each end so when a user dials the local 7 digit number for the other end it will try to rute the call through the IAX2 trunk before trying the PSTN lines. When the call comes in on the other end I want it to hit my external IVR. The IAX2 trunk connection is working great a call going to 1234567 goes over the Internet to the other end but then on the receiving end it tries to dial out a zap channel to call back in the 1234567 zap channel. I have the outbound route to match 1234567 to the IAX2 trunk and in the IAX2 trunk I need to strip off the 7 digit number and replace it with a *02 to call my external IVR on the other end. For testing I have been trying to make the call connect to my extension on the other end but I am not having any luck. I need some help to make this Dial Rule work. This is what I am trying to use 1234567|+219 From what I have read this should strip off the exact matching 7 digit local number 1234567 and add a prefix of 219 but it does not. From trying different orders and mixing this around I am only able to do one or the other. I can either strip off the 1234567 which does nothing or I can add a 219 prefix that calls 2191234567 on the other end. What do I need to do to strip off the 7 digit number that was dialed and replace it with a 219 or replace it with a *02? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] Trying to make SIP calls through Asterisk with anonymous connection
I am trying to setup SIP to SIP calling between Asterisk managed networks. I want to make it so that people can call SIP:[EMAIL PROTECTED] and they connect to my Asterisk and get my external IVR then they can dial my extension or navigate extensions just like they would if they had called using a PSTN line. I also want to call other people using my Asterisk and dialing an external SIP like so SIP:[EMAIL PROTECTED] I want out going SIP calls to me managed by my Asterisk so I can transfer them to other people in my office or conference or use any of the other great features that Asterisk provides. I do not want to go SIP direct to SIP, I want to go SIP to Asterisk to Asterisk to SIP and connect to the far end Asterisk without requiring me to register my Asterisk server with the far end Asterisk server. For testing I have setup two servers running Asterisk. Both are on the Internet with static IP addresses and behind firewalls. The firewalls are configured to allow TCP UDP ports 5060 to 5082 and 10001 to 2 to connect directly to the Asterisk servers. This allows SIP and RTP connections from the outside. I have tested with Twinkle (a Linux softphone) and can connect to a registered account with NAT from external IPs. I have also set the Asterisk servers to allow incoming anonymous SIP calls to connect to the from-external. When I try to dial SIP:[EMAIL PROTECTED] Asterisk tries to dial the some_extension on my local network not the other network. I reconnect to the running asterisk using -r and watch when I dial and it does not report the @other_url only the some_extension. I am not having much luck finding the documentation I need. Can someone point me to a How-To on doing this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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
