RE: [Asterisk-Users] SIP phone failover using DNS SRV?
Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV? Definitely we have been doing this for quite a while. If so, which phone worked for you? I'm assuming you set up your DNS SRV records so that the IP addresses of A and B are associated with the same name, and both servers have equal priority and equal weight. We have this working great with both Polycom and Sipura devices. We have servers of different priority (i.e. - primary then failover). The name is the same. In order to make calls through B after A goes down, do you have to wait as long as the registration retry interval? Or can you make calls through B as soon as you pick up the phone and dial, because the INVITE message through A fails, and the phone re-sends the INVITE through B? The way we have it working A is a higher priority than B, so every phone will register to A unless it is down (or the phone is having connectivity issues and points itself to B automatically). In this case when A goes down each phone will automatically failover to B when the next call is placed. There is still an issue of inbound calls, but most carriers will provide a mechanism to fail calls over as well (if the server is truly down). The regisration retry interval (to the best of my knowledge) is how long the phone will wait before attempting to re-register with servers of a higher priority after failing over. We tend to set this as a pretty low number because we want things to get back to the primary as soon as possible (since our carriers will fail back immediately for any new call coming in once the server is back up). -Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage Linksys Router - Life after Vonage
I too have heard of people persuading a vonage tech to provide the password to log into and factory reset their device, but I get the impression that it is an uncommon occurrence.. you'd be lucky, basically. I have an ATA-186 that Vonage unlocked for me. They used to just charge $20 or so (on top of the cancellation fee) to unlock any device that wasn't active. You'll have to get through their first line of support and eventually transferred. They *may* have changed the policy with the linksys boxes, as they couldn't do it when they first came out (I tried). Now that linksys sells an unlocked version they may well be willing though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using as FAX 100% IP
I'd say that would depend on the configuration you are considering. We have a number of fax machines running off of sipura spa-2000's that connect to a remote asterisk server and terminate to the pstn via voip as well. I'd say it's about 90% reliable at this point. However, we've noticed quite a bit of variability around the quality of the connection and underlying provider you are using for termination - so your results will vary. It also seems a couple of other tricks help: (1)put an ADSL filter at the fax machine end, this seems to help filter out noise from the signal and slow down the analog modem (not sure if this is an old maid's tale or not, but seems to help). (2)if you are using the sipura as your ATA, disable all of the fax detection. Just make sure you are only allowing ULAW as your codec. Other than that it's pretty straightforward. If you are looking to do something to send receive to files, etc. we haven't had any luck getting something along those lines to work w/pure voip. -Steve Hakem Taourchi wrote: Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? If yes using which components please? Any help is greatly appreciated ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Using as FAX 100% IP
I don't believe this will work, but haven't tried myself. TAFM requires spandsp. I'd do some investigation there as to whether spandsp can function with g711 (the last i checked it didn't). Good luck - if you make any progress please post to the list. -Steve Hakem Taourchi wrote: Thank you ver much for this help Steven What I am planning is this: 1-) Receive fax on a DID that is being routed in IP to the asterisk server; 2-) Based on the rule on that incoming fax, Asterisk needs to capture it, store it as pdf file and e-mail it to a predefined destinoatin (based on DID); 3-) If 2 is not possible, then send fax to PSTN destination using voip; Did you use this http://tafm.sourceforge.net/ to make ATA sepura work ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steven Kokinos Envoyé : mercredi 23 février 2005 15:46 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Using as FAX 100% IP I'd say that would depend on the configuration you are considering. We have a number of fax machines running off of sipura spa-2000's that connect to a remote asterisk server and terminate to the pstn via voip as well. I'd say it's about 90% reliable at this point. However, we've noticed quite a bit of variability around the quality of the connection and underlying provider you are using for termination - so your results will vary. It also seems a couple of other tricks help: (1)put an ADSL filter at the fax machine end, this seems to help filter out noise from the signal and slow down the analog modem (not sure if this is an old maid's tale or not, but seems to help). (2)if you are using the sipura as your ATA, disable all of the fax detection. Just make sure you are only allowing ULAW as your codec. Other than that it's pretty straightforward. If you are looking to do something to send receive to files, etc. we haven't had any luck getting something along those lines to work w/pure voip. -Steve Hakem Taourchi wrote: Hello, Before putting any effort, I would like to know if somebody has successfully run asterisk receiving FAXs in IP and sending them out in IP as well? If yes using which components please? Any help is greatly appreciated ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] All you polycom folks.....
Does anyone know where to obtain the 2.5 / 1.3.1 bootrom/app? My account on polycom's site keeps pointing me at documentation only. Regards, -Steve On Sep 1, 2004, at 10:00 AM, Matthew Marlowe wrote: I'm using the same SIP version, everything is running great except as I've said before that setting the default ring type isn't working and incoming calls only displays name and not name and number.. From: [EMAIL PROTECTED] on behalf of Reid A. Forrest Sent: Wed 9/1/2004 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] All you polycom folks. I'm using SIP 1.3.1 with Boot RPM 2.50 and so far everything's running great. I don't use # transfer though, so haven't tried that. I use the softkeys instead to transfer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Wednesday, September 01, 2004 8:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] All you polycom folks. I left my phone at home I think Im using sip 1.3.1.. It's 1.3. something Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: Tuesday, August 31, 2004 10:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] All you polycom folks. Just out of curiosity, What version of CVS and Polycom SIP software are you running happily? Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0? I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with poor results. Transferring did not work as expected. Using the # key to do blind transfers after a call was on hold did not work. Just curious. Thanks, - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID's in the Czech Republic
Does anyone know of any provider(s) that can provide DID's for the Czech Republic? Regards, -Steve
Re: [Asterisk-Users] Sipura-SPA2000 background noise
I too have the same problem on a few units, but not on others. I also have been having difficulty hooking up multiple lines from one Sipura to the same multi-line phone system (seems to create a line cross) but have no problems with either cisco or dlink boxes. In general they are nice units, but I suspect they may have had a batch go out that were noisy. -Steve On Jun 1, 2004, at 3:10 PM, [EMAIL PROTECTED] wrote: I hear the exact same noise on 2 units I purchased a few months ago. I've been in contact with sipura support and they are willing to try RMA'ing one of my units. As soon as I can get to the site with the sipura, I'll be sending it in. I'll post my results to the list. btw, I'd have to agree that its not comfort noise, its very similar (only much louder) to the hiss that the old digium fxs modules had on the tdm boards. Mark At 10:21 AM 6/1/2004, you wrote: Not really a comfort noise. I say anything and it doesent go away. It sounds like a shielding issue. I have tried to relocate the unit but it doesn't seem to help. -Original Message- From: Kevin Walsh [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 11:46 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura-SPA2000 background noise Kevin [EMAIL PROTECTED] wrote: I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I noticed there is a fair amount of background white noise that is noticeable, especially after breaking the dial tone. After pressing a '1' to break the dial tone, there is a fair amount of noise that is evident. I do not notice this condition on the Cisco ATA's. I plugged the Sipura in the same location as the Cisco ATA. Anyone else have this condition with the Sipura? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco ata-186 behind NAT
i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions, i've been able to get the device to register to asterisk, and make calls without any problem. however, the asterisk box cannot see the adapter, and does not respond to hangup requests (therefore it would seem that the rtp stream is working properly in both directions, but SIP traffic is not finding it's way back). i have been focusing on two parameters in an attempt to get things functioning normally - namely NatTimer and ConnectMode. I have the following settings currently: ConnectMode: 0x20460400 (have also tried what i've seen elsewhere - 0x00460400, and 0x01a40400) NatTimer: 0x0054000a I've also tried the defaults and anything else suggested by others. If anyone has an ATA-186 running in a similar configuration and could share their configs with me it would be greatly appreciated. Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY re quests)
Beyond this, you can still just use the NAT keepalive in the Sipura. While It only provides for either a NOTIFY or REGISTER (which both generate errors in asterisk) if you change it to something else (I just have it send blank, but a few ... or anything will do) asterisk won't complain and the data is sent every few seconds, keeping the firewall open. I've also found setting the register to something low (I use 300s) also helps when you do have to use qualify, in case asterisk loses the connection the device will only be offline until the next register. -Steve On May 22, 2004, at 3:32 PM, Darren Nay wrote: Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Saturday, May 22, 2004 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY requests) At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: [snip] Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O Do you have difficulties with the Sipura SPA-2000 (or other Sipura boxes) and Asterisk? I've found no problems, even behind NAT, though I have only tried behind one or two NAT devices (OpenBSD and Apple Airport.) It's surprising that Sipura doesn't include STUN as an option - their list of options is so huge that I always assumed I had just missed it, but now that I look closer I suppose you're right. Do Asterisk users even really need STUN? I've never found it to be required after the NAT issues were worked out of Asterisk... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom ip 500 registration problems
hello all, I'm having problems getting my polycom soundpoint ip 500 working, and was wondering if anyone would be willing to share their config files with me (the polycom configs). I have managed to get my boot server up and running, and the phone successfully updated its ROM, and downloaded the config files i have put together for it (the display shows correctly but the line won't register). in asterisk sip.conf i have a usual configuration: [polycom1] type=friend host=dynamic context=home secret=mypasswd callerid=Polycom Line1 nat=yes canreinvite=no mailbox=2400 qualify=yes Any help is greatly appreciated. regards, -steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failover Scenario - synchronizing voicemail key files
I currently have several asterisk servers geographically distributed (for automatic fail-over in the event of either a network or server problem). My carrier delivers to each server based on the same priorities that I have set inthe DNS SRV records which the clients point to. Users always have dialtone regardless of a single server failure. In addition, once they have re-registered with the backup server they will receive calls as normal until the primary becomes available again. However, there is a slight issue in a few circumstances (and others not listed): (1)The carrier can see the primary, but the clients can't (or vice versa) meaning that a user won't receive the call (butcan make calls, andinboundwill go into voicemail). (2)A user receives a voicemail while temporarily failed-over to another server, then re-registers with the primary server (meaning the voicemail will be "stuck" on the secondary server). The solutionI amusingis to run rsync to synchronize configuration files, and (hopefully soon)voicemail between all of my servers. The config files are no problem since changes will always be made on the primary, then propagated out (I run a script to look at the checksum of /etc/asterisk every 5 minutes, if it is different i execute a reload). However, the voicemail (and potentially sound files, etc.) are a different story. Best I can tell, voicemail data is entirely contained within /var/spool/asterisk/voicemail. Are there any other files/directories I should be concerned with? Discussion around this topic in general appears to be somewhat spotty (and there's nothing on voip-info.org) so any comments or suggestions people have would be appreciated. -Steve
[Asterisk-Users] zaprtc
does anyone out there using zaprtc know how to go about initializing it at boot time? i have it compiled and working properly, but there is very limited documentation. -Steve
RE: [Asterisk-Users] zaprtc
Fran, Thanks for the message. In between my original message and your response I actually did something a bit different. From within /etc/init.d/asterisk (which I call from chkconfig) I added the following lines: start) echo -n Starting Asterisk PBX: #/sbin/modprobe ixj /sbin/modprobe zaptel /sbin/insmod zaprtc /sbin/rtcsetup daemon /usr/sbin/safe_asterisk RETVAL=$? echo [ $RETVAL -eq 0 ] touch /var/lock/subsys/asterisk ;; stop) echo -n Shutting Asterisk PBX: killproc safe_asterisk killproc asterisk killproc rtcsetup #/sbin/rmmod -r ixj /sbin/rmmod -r zaptel /sbin/rmmod -r zaprtc RETVAL=$? echo [ $RETVAL -eq 0 ] rm -f /var/lock/subsys/asterisk ;; This works for me as well, and has the added bonus (at least in my case) of keeping the asterisk related items more self-contained. -Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fran Boon Sent: Tuesday, April 20, 2004 6:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] zaprtc On Tue, 2004-04-20 at 17:23, Steven Kokinos wrote: does anyone out there using zaprtc know how to go about initializing it at boot time? i have it compiled and working properly, but there is very limited documentation. Yup, works great for me :) Add this to rc.local to get it initialised at boot: insmod /lib/modules/2.4.21-9.ELcustom/misc/zaprtc.o /usr/local/bin/rtcsetup (Obviously modify the kernel path if required - this is for RHES3) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stable from 4/20 launching many processes
i have a quick question from the latest build in the stable branch. in all of the previous builds of asterisk i have used, calling either asterisk itself or safe_asterisk spawns one asterisk process, like this: root 11218 0.0 0.1 5244 936 pts/0S20:55 0:00 /bin/sh /usr/sbin/safe_asterisk root 11220 3.0 1.0 152900 4876 pts/0 S20:55 0:00 asterisk -vvvg -c however, in the latest build, i am seeing the following behavior (tested both starting manually and with safe_asterisk): root 797 0.0 0.2 4248 1136 ?S23:52 0:00 /bin/sh /usr/sbin/safe_asterisk root 799 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 800 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 801 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 802 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 803 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 821 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 823 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 824 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 825 0.0 0.9 102620 4952 ? R23:52 0:00 asterisk -vvvg -c root 826 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 827 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 830 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 831 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 832 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c root 833 0.0 0.9 102620 4952 ? S23:52 0:00 asterisk -vvvg -c which is exactly 15 instances of asterisk. this is certainly a usual way of running for many different applications, but i was not aware asterisk was one of them. i would think there was something haywire going on, however, if i start a single instance of asterisk, then stop it gracefully, all processes do indeed stop. Is this expected behavior, or something unexpected that i should be concerned with? Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy problems (was music on hold problems)
I definitely have the correct usb card. i have another machine running fedora core 1 (kernel 2.4.22) and everything works no problem. i remember reading somewhere that there were some driver updates in 22 that resolved some problems. seems like I could be running into that. -steve On Apr 15, 2004, at 6:52 PM, Iain Stevenson wrote: --On Thursday, April 15, 2004 6:43 pm -0400 Steven Kokinos [EMAIL PROTECTED] wrote: Actually, after rebooting my machine music on hold started working properly. Not sure what the issue was. As for ztdummy, I am having a more substantive issue with that, which is keeping me from getting meetme working. snip however, usb-uhci.o does in-fact exist. Does anyone have any thoughts? You have an appropriate USB card installed? - ztdummy won't work with ohci cards. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning from Asterisk
I have had problems with the NOTIFY packet being sent (not problems, just annoying warnings) when using keepalive with the Sipura SPA-2000. Asterisk will complain both using the Register and Notify methods (which are the two that the Sipura uses). The NOTIFY warning isn't actually causing any trouble, but I opted to just send no data (a carriage return will also work) to get the adapter to keep the port open. Sending blank info keeps asterisk from complaining as well. If you having this problem otherwise it is certainly something else. -Steve On Apr 16, 2004, at 1:20 PM, Kurt wrote: On the * console do a sip debug and look for the notify packet. It should give you a reason why it is being sent. Kurt __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] external voicemail access - solved (mostly)
thanks to those who replied. I have managed to get the functionality I
was looking for working with a series of Macros. However, it doesn't
work as simply as I would like. There are two issues I've run into:
(1)Goto provides no way to pass variables between one context and
another.
(2)I can't find any way to Goto a specific point within a Macro when
calling it.
Mostly this is a result of the background command listening for
extensions in the current context. If background is run from within a
Macro, then it will terminate the macro and return to the current
context to execute whatever user input was just captured. In order to
get the behavior I'm looking for (user calls into voicemail, presses *
to be prompted for a password and check messages, press # to skip the
greeting and leave a message) I had to have 3 macros:
(1)vm - leave voicemail
(2)vm-nogreet - simply provide a beep
(3)checkmessage
Ideally this would be one larger macro, where the starting point could
be specified as well as passing the arguments along.
[macro-vm]
exten = s,1,Answer
exten = s,2,Background(${VMAILPATH}/${ARG2}/${ARG1}/unavail)
exten = s,3,VoiceMail2(s${ARG1})
exten = s,4,Hangup
[macro-vm-nogreet]
exten = s,1,Answer
exten = s,2,VoiceMail2(s${ARG1})
exten = s,3,Hangup
[macro-checkmessage]
exten = s,1,VoiceMailMain2(${ARG1})
exten = s,2,Hangup
In the inbound context I do the following (xxx is the rest of the
phone number):
[line-in]
exten = xx1638,1,Dial(${P1}${P2}${P3},25,Tr)
exten = xx1638,2,Macro(vm,${P1_VM},${P1_VM_CONTEXT})
exten = xx1638,3,Hangup
exten = *,1,Macro(checkmessage,${P1_VM})
exten = i,1,Macro(vm-nogreet,${P1_VM},${P1_VM_CONTEXT})
This does everything in a fairly general way. However, if anyone knows
how to address my points above this could be done much cleaner in a
single macro (and if there is a way to keep background isolated within
the macro it would be easier still). One note - this will actually
interrupt the greeting and send to the beep for voicemail regardless
of what the user presses, as long as it isn't the * key. Since
background will always bounce to here I thought it would be better to
force someone to leave voicemail than get a fast busy with an
inadvertent button press.
-Steve
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[Asterisk-Users] music on hold problems
i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. I have the following in extensions.conf: exten = 2100,1,Answer exten = 2100,2,MusicOnHold(default) and have uncommented the default line in musiconhold.conf: [classes] default = quietmp3:/var/lib/asterisk/mohmp3 I also installed mpg123, and created a symlink to /usr/bin (which is where it seems asterisk looks for it). Does anyone have any idea as to what I'm doing wrong here? Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy problems (was music on hold problems)
Actually, after rebooting my machine music on hold started working properly. Not sure what the issue was. As for ztdummy, I am having a more substantive issue with that, which is keeping me from getting meetme working. while ztdummy compiles cleanly, i can't actually get it to load properly. [EMAIL PROTECTED] root]# modprobe ztdummy /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed however, usb-uhci.o does in-fact exist. Does anyone have any thoughts? Regards, -Steve On Apr 15, 2004, at 5:27 PM, Tony Mountifield wrote: In article [EMAIL PROTECTED], Steven Kokinos [EMAIL PROTECTED] wrote: i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. Music on hold requires a zaptel timing source. If you do not have any zaptel cards in your system, you will need to install either ztdummy (only if you have a uhci type of USB) or zaprtc. See http://www.voip-info.org/tiki-index.php?page=Asterisk+timer Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] background / goto commands
I'm working on setting up a macro that will allow users to call their
own DID number, and when they hear their voicemail greeting hit the *
key and be prompted for their password to check vmail.
For some reason though the background command isn't working as I'd
expect it to:
[macro-vmessage]
exten = s,1,Answer
exten =
s,2,Background(/var/spool/asterisk/voicemail/sixthree/${ARG1}/unavail)
exten = 1,*,Macro(checkmessage,${ARG1})
exten = 2,*,Hangup
exten = s,3,VoiceMail2(s${ARG1})
exten = s,4,Hangup
[macro-checkmessage]
exten = s,1,VoiceMailMain2(${ARG1})
exten = s,2,Hangup
If I do nothing during the greeting all behaves as I would expect (user
is given the unavailble greeting, then a beep to leave the message) and
the message is delivered properly.
However, if I hit the * key during the greeting, I get the following
error:
== Spawn extension (macro-vmessage, s, 2) exited non-zero on
'SIP/2400-9055' in macro 'vmessage'
Apr 14 17:43:06 WARNING[1192437440]: pbx.c:1825 ast_pbx_run: Invalid
extension, but no rule 'i' in context 'home'
Can anyone point me in the right direction as to what I'm doing wrong
here?
Regards,
-Steve
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Re: [Asterisk-Users] background / goto commands
Already tried that, doesn't do anything.
It apears what is happening is that Background interprets any key as
the beginning of a user dialing an extension from the default context
of that extension, so invariably it jumps out of the macro into the
main context.
Can't figure out how to keep it local to the macro.
-Steve
On Apr 14, 2004, at 6:11 PM, Chris A. Icide wrote:
On 02:44 PM 4/14/2004, Steven Kokinos wrote:
I'm working on setting up a macro that will allow users to call their
snip
exten = 1,*,Macro(checkmessage,${ARG1})
exten = 2,*,Hangup
try:
exten = *,1,.
exten = *,2,.
instead.
snip
Can anyone point me in the right direction as to what I'm doing wrong
here?
Regards,
-Steve
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[Asterisk-Users] RTP Read error
I've been intermittently seeing the following warning: Apr 14 18:35:04 WARNING[1192437440]: rtp.c:386 ast_rtp_read: RTP Read error: Resource temporarily unavailable which doesn't appear to have any effect on the current call, but isn't anything I've seen before either. Any thoughts on whether this is something to be concerned with? Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange SIP behavior w/NAT Keepalive
Hello- I have several Sipura SPA-2000's at different locations (all behind Linksys WRT54G boxes). When setting: nat=yes qualify=yes Things work properly about 90% of the time, however, if a remote end loses the connection briefly, then asterisk can't see the adapter until the next registration attempt. Outbound calls will still function, but inbound obviously won't since asterisk can't see the remote client anymore. The better way (in my book anyway) is to enable keepalive on the remote end, so that it can keep the hole open in the NAT. However, if I enable this with the Sipura using the NOTIFY method, I get the following error: Apr 10 23:24:25 NOTICE[1116941120]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from 'xx.xxx.xx.143' And if I do the same with the REGISTER method, I get a different issue: Apr 10 23:25:12 NOTICE[1116941120]: chan_sip.c:3461 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway -- Got SIP response 404 Not Found back from xx.xxx.xx.143 However, strangely, if I deviate from either of these two and put in $OPTION in the Sipura (which is something it doesn't recognize as a keepalive variable - it just blankly passes it along) Asterisk does in fact keep the port open, ad doesn't throw off a notice. Upon setting sip debug, I get the following: Sip read: $OPTIONS 1 headers, 0 lines But only in debug mode. This does in fact keep the connection open, but seems like a pretty ugly hack to me. Does anyone have any suggestions here, it would seem that Asterisk should recognize an inbound NOTIFY request (it sends them out itself)? Thanks, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] syslog error
Hello, I have been running into a problem on my server (which I believe was the cause of an O/S crash earlier today). I am consistently seeing the following messages in /var/log/messages: Apr 8 05:24:04 east insmod: /lib/modules/2.4.20-8/misc/torisa.o: insmod char-major-196 failed Apr 8 05:24:04 east kernel: Zapata Telephony Interface Registered on major 196 Apr 8 05:24:04 east kernel: No ISA tormenta card found at d Apr 8 05:24:04 east kernel: Zapata Telephony Interface Unloaded Apr 8 05:24:04 east kernel: Zapata Telephony Interface Registered on major 196 Apr 8 05:24:04 east kernel: No ISA tormenta card found at d Apr 8 05:24:04 east kernel: Zapata Telephony Interface Unloaded Apr 8 05:24:04 east insmod: /lib/modules/2.4.20-8/misc/torisa.o: init_module: Input/output error Apr 8 05:24:04 east insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg I have put the following line in modules.conf (under [modules]): noload = chan_zap.so And I am also receiving the following error when doing a modprobe of ztdummy: /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed Does anyone have any ideas on why this might be happening? It looks to me like either something is missing on my system or I did something incorrectly at compile time. Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata required?
Hello- As part of the asterisk build/installation instructions it mentions that the zaptel drivers should be built and configured first. My question is whether they are required at all, in the case of a system with no hardware cards at all (as is the situation in my case). With them loaded I continually get the following message on my console (server not asterisk): Zapata Telephony Interface Registered on major 196No ISA tormenta card found at dZapata Telephony Interface Unloaded which seems logical given that I don't have any zap hardware. how would i go about unloading this module and/or does it need to be compiled at all in this case? Regards, -Steve
[Asterisk-Users] External access to voicemail
in my setup i have several users with DID lines coming in from various sip/iax providers. within our old phone system, a user could call their own DID line, then hit the * key when they hear their voicemail greeting and be prompted for their password. is there any way this could be replicated within asterisk? i'm having trouble figuring it out since it steps through things sequentially, whereas i want to scan for input during the playback. any help would be greatly appreciated. regards, -steve
[Asterisk-Users] Question receiving calls via SIP
Hello-
I am in the process of adding a new provider to my asterisk box (both
for outbound termination as well as inbound DID). They are going to be
delivering and receiving traffic via SIP only.
Now, in IAX via Voicepulse or others I know that I can simply have one
registration statement along with an inbound context, then in
extension.conf map the outbound context.
from iax.conf:
register = in-:[EMAIL PROTECTED]
from extensions.conf
[voicepulse-in]
exten = 212xxx,1,Dial(${PHONES1}${PHONES2},30)
exten = 212xxx,2,Voicemail2(u${PHONES1VM})
exten = 212xxx,3,Hangup
I know this way I only have to register once, but can receive calls on
several inbound DID numbers without any problem, provided they are all
mapped similar to what I have above within extensions.conf.
My question is whether or not the same thing will work for a sip
provider, as it will be pretty cumbersome from a networking standpoint
to have a registration statement for each DID (as opposed to simply
having a new extension statement). In the syntax of the sip
registration statement it appears to always end with the /extension
that is supposed to be associated with that account.
Can the /extension be left off of the statement in sip.conf, and picked
up in the same way DID's are used in iax.conf? If not, why are there
separate methods for registering each protocol, would seem cleaner to
have a consistent way of dealing with this.
Regards,
-Steve
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[Asterisk-Users] problems getting inbound to work @ voicepulse
Hello- I'm obviously doing something wrong here in trying to get an inbound DID to work with voicepulse. I have an outbound context set-up for those calls in iax.conf, and the appropriate register in- statement. within extensions.conf I am doing something like this: exten = 212xxx,1,Dial(SIP/admin,t) (where admin is the phone i am looking to forward to from sip.conf). i'm getting the following errors from iax2 debug: Apr 2 16:00:54 NOTICE[1133718080]: chan_iax2.c:5087 socket_read: Rejected connect attempt from 66.xxx.xxx.xxx, request '[EMAIL PROTECTED]' does not exist Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00034ms SCall: 4 DCall: 00233 [66.234.228.132:4569] CAUSE : No such context/extension Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00034ms SCall: 4 DCall: 00233 [66.xxx.xxx.xxx:4569] CAUSE : No such context/extension Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 00233 DCall: 4 [66.xxx.xxx.xxx:4569] Any ideas would be greatly appreciated. I'm not sure if I need to put something specific in for the inbound number in sip.conf, or extensions.conf. The instructions in the howto and on voicepulse both seem somewhat vague. -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems getting inbound to work @ voicepulse
Scott- Thanks for the tip. It was in fact that I didn't have the two contexts matching. Once I resolved that everything is working great. -Steve On Apr 2, 2004, at 6:21 PM, Scott Laird wrote: On Apr 2, 2004, at 3:12 PM, Steven Kokinos wrote: Hello- I'm obviously doing something wrong here in trying to get an inbound DID to work with voicepulse. I have an outbound context set-up for those calls in iax.conf, and the appropriate register in- statement. within extensions.conf I am doing something like this: exten = 212xxx,1,Dial(SIP/admin,t) Which context is this in? You need to have a 'context=something' line in the VoicePulse block in iax.conf that points to the context in extension.conf that has this line. I'm using NuFone over IAX with an 866-xxx DID, and it works just fine. See http://www.voip-info.org/tiki-index.php?page=NuFone%20Settings for an example; it looks more readable then the VoicePulse example. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem connecting to voicepulse don't know how to authenticate
Hello, I've just compiled and configured my asterisk box, and so far things have gone pretty smoothly. Now I am working on being able to make a call via voicepulse, and have run into an issue. When I connect via my sip phone (using xlite) and dial a pstn number I can see that it is being directed out properly (i have voicepulse as the default). However, I am getting the following message: Executing Dial(SIP/admin-c5d2, IAX2/[EMAIL PROTECTED]/1xx) in new stack -- Called [EMAIL PROTECTED]/1xx Apr 1 16:03:18 WARNING[1133718080]: chan_iax2.c:5061 socket_read: I don't know how to authenticate to xx.xxx.xxx.xxx (replaced real information with x's). Any thoughts would be greatly appreciated. Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem connecting to voicepulse don't know how to authenticate
actually figured this one out on my own, had incorrectly labeled secret as password in iax.conf. On Apr 1, 2004, at 6:14 PM, Steven Kokinos wrote: Hello, I've just compiled and configured my asterisk box, and so far things have gone pretty smoothly. Now I am working on being able to make a call via voicepulse, and have run into an issue. When I connect via my sip phone (using xlite) and dial a pstn number I can see that it is being directed out properly (i have voicepulse as the default). However, I am getting the following message: Executing Dial(SIP/admin-c5d2, IAX2/[EMAIL PROTECTED]/1xx) in new stack -- Called [EMAIL PROTECTED]/1xx Apr 1 16:03:18 WARNING[1133718080]: chan_iax2.c:5061 socket_read: I don't know how to authenticate to xx.xxx.xxx.xxx (replaced real information with x's). Any thoughts would be greatly appreciated. Regards, -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nuvio users?
There are a lot of wholesale VOIP providers now, they are probably just buying capacity from one of them and either re-branding, or are using a custom gateway and/or commercial solution. -Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zac Amsler Sent: Wednesday, March 24, 2004 9:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Nuvio users? It kinda looks like they are reselling Vonage service.. Zac -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Tuesday, March 23, 2004 11:10 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Nuvio users? Hi, Anyone gotten Asterisk to work with Nuvio (http://www.nuvio.com)? They do support SIP phones (if you give them the MAC address)... Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
