[asterisk-users] Asterisk freezes with Fixup failed on channel SIP/...MASQ
On a production system, running 1.4.17 (compiled from bristuff-0.4.0-test6-xr1) we had this strange issue two times in the last weeks: [2009-01-13 13:58:30] WARNING[1213] channel.c: Fixup failed on channel SIP/2332-081d0108MASQ, strange things may happen. [2009-01-13 13:58:30] WARNING[1213] channel.c: Hangup failed! Strange things may happen! [2009-01-13 13:58:30] WARNING[1213] channel.c: Failed to perform masquerade [2009-01-13 13:58:30] WARNING[1213] channel.c: Channel 'SIP/2332-081d0108' may not have been hung up properly and: [2009-01-23 14:27:17] WARNING[21528] channel.c: Fixup failed on channel SIP/2332-083c3778MASQ, strange things may happen. [2009-01-23 14:27:17] WARNING[21528] channel.c: Hangup failed! Strange things may happen! [2009-01-23 14:27:17] WARNING[21528] channel.c: Failed to perform masquerade [2009-01-23 14:27:17] WARNING[21528] channel.c: Channel 'SIP/2332-083c3778' may not have been hung up properly Both times all SIP channels got stuck and the CLI became inresponsive. Calls continued for a while, but new SIP calls could not be established. On the second time this happended, all SIP phones could not subscribe to the Asterisk any longer and a few minutes later the log filled with: [2009-01-23 14:43:21] ERROR[22319] chan_sip.c: Call to peer '2333' rejected due to usage limit of 10 On the CLI one could see, that there were 100s of (rejected) calls to this SIP phones. The phones that show up in the ERROR messages are in a group call made by a Dial(Local/...Local.../Local/...) construct. But other SIP phones were affected as well. It seemed like the whole chan_sip module became stuck. I also could not unload chan_sip.so, but can't remeber the exact error message it gave. The only thing that was left was to restart Asterisk. Can someone give me some clue what the 'Fixup failed ...' and 'masquerade' warnings actually mean? Any help appreciated. Udo ___ -- 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] [SOLVED] Nortel IP phone i2002 - DHCP server unreachable
Perhaps this would help... http://blog.michaelfmcnamara.com/2007/10/dhcp-options-voip/ Gives details on the dhcp option string needed for the phones and explains that without it the phone will not accept a DHCP response... d 2009/1/24 Joseph syscon...@gmail.com Thanks for the input. Yes, I have all the correct setting in the phone. However, it turn out that I need to have as DHCP server Nortel BCM (Business Communication Manager), whatever it is. It must be some proprietary stuff. So the only option for me was to setup IP manually; and it did work following this guide: http://www.oneconnect.ca/files/userguide.i2002-i2004ConfigurationInstructions.pdf -- #Joseph GPG KeyID: ED0E1FB7 On Fri, 23 Jan 2009, Alexander Lopez wrote: 1 Can you verify that you have a DHCP server running on that network segment? 2 Can you verify that the Ethernet port on the phone is indeed seeing link from the switch? 3 Have you run wireshark/tcpdump to see if anything is traveling to/from the phone? Alex -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Joseph Sent: Friday, January 23, 2009 9:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nortel IP phone i2002 - DHCP server unreachable Is anybody using Nortel IP Phone? I have (second hand) Nortel i2002 phone and when it boots I get: DHCP server unreachable F/W version: 0604D9C My setting: DHCP? [0-No, 1-Yes]: 1 DHCP: 0-Full, 1-Partial: 0 Can any body suggest how to troubleshoot it? ___ -- 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] local dialing
wich limitations?
why you dont just answer the incoming calls in TEST context?
give mucho more info so we can help you.
David
2009/1/24 Pezhman Lali pezhman_l...@yahoo.com
Dear,
because of using dial(local/...) each incoming calls (_12X.) makes 4 ports
on asterisk.
I can not use goto , because of some limitations.
is any way to decrease it?
Best,
[MAIN]
exten = _12X.,Dial(LOCAL/${ext...@test/n,60)
[TEST]
exten _X.,1,Dial(${ext...@next_gateway,60)
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Re: [asterisk-users] registration problem using asterisk 1.6
2009/1/22 Laurent Bonny laurent.bo...@gmail.com Hello, I am trying to connect an asterisk 1.6 to a trunking plate forme. With asterisk 1.4.x I added to sip.conf a line asking for registration in the form of: register = xx...@domain.com:Password:xx...@domain.comassword%3axx...@domain.com @domain.com Unfortunately, as you can see, my usernames have to be of the form xxx...@domain.com which means that I had to put 2 @ at the end of my line, 1 for the username and 1 for the domain. In Asterisk 1.6 it doesn't seems to work anymore the @ being a reserved sign, and something like this line being impossible. Is it a bug from asterisk (as I don't see why I couldn't have a username in this form)? Would you know of a way to register my users correctly? Could you not do something like this? register = X:passw...@provider [provider] type=peer host=domain.com fromdomain=domain.com username=X fromuser=X secret=password Then it would register as xx...@domain.com... d ___ -- 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] Logging outgoing calls
and what about add a custome field or setup a variable on outgoing calls and use the common cdr and then filtering by that field. David 2009/1/24 Tilghman Lesher tilgh...@mail.jeffandtilghman.com On Friday 23 January 2009 18:22:16 Pascal Bruno wrote: Is it possible to log just the outgoing calls using cdr_odbc into a custom mysql database table? my table will look like this: | call_status | |-- --| | · id | | · destination | | · status | || I just need to store the destination number and the status of the channel for example BUSY, UNAVAILABLE etc... Yes, if you install the cdr_adaptive_odbc backport. See the sample config file for more information. Web: http://svncommunity.digium.com/view/tilghman/branches/1.4 SVN: http://svncommunity.digium.com/svn/tilghman/branches/1.4 -- Tilghman ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME are equipped today with PBX connected with TE/PtMP interfaces to telco BRI lines. When we address those SME, my opinion is that it's very useful to be able to support any combination of TE/NT, PtP/PtMP modes. Latest 1.6 Asterisk and 1.4.8 Libpri introduced a new set of welcomed ISDN BRI features. Unfortunately, NT/PtMP is not available at this time, in latest Zaptel/Asterisk/Libpri. My question is what is the policy concerning NT/PtMP ? Is it really hard to extend Libpri to support this mode ? Or shall mISDN remain the way to go when NT/PtMP is needed ? Regards ___ -- 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 freezes with Fixup failed on channel SIP/...MASQ
Copy paste from freeswitch.org Asterisk uses a modular design where a central core loads shared objects to extend the functionality with bits of code known as modules. Modules are used to implement specific protocols such as SIP, add applications such as custom IVRs and tie in other external interfaces such as the Manager Interface. The core of Asterisk is a threading model but a very conservative one. Only origination channels and channels executing an application have threads. The B leg of any call operate only within the same thread as the A leg and when something happens like a call transfer the channel must first be transferred to a threaded mode which often times includes a practice called channel masquerade, a process where all the internals of a channel are torn from one dynamic memory object and placed into another. A practice that was once described in the code comments as being nasty. The same went for the opposite operation the thread was discarded by cloning the channel and letting the original hang-up which also required hacking the cdr structure to avoid seeing it as a new call. One will often see 3 or 4 channels up for a single call during a call transfer because of this. /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ This became the de facto way to pull a channel out of the grips of another thread and the source of many headaches for application developers. This uncertain threading scheme was one of the motivating factors for a rewrite. Asterisk uses linked-lists to manage its open channels. A linked-list is a series of dynamic memory chained together by using a structure that has a pointer to its own type as one of the members allowing you to endlessly chain objects and keep track of them. They are indeed a useful programming practice but when used in a threaded application become very difficult to manage. One must use mutexes, a kind of traffic light for threads to make sure only 1 thread ever has write access to the list or you risk one thread tearing a link out of a list while another is traversing it. This also leads to horrible situations where one thread may be destroying or masquerading a channel while another is accessing it which will result in a Segmentation Fault which is a fatal error in the program and causes it to instantly halt which, of course means in most cases all your calls will be lost. We've all seen the infamous Avoiding initial deadlock message which essentially is an attempt to lock a channel 10 times and if still won't lock, just go ahead and forget about the lock. 2009/1/24 Udo Schacht-Wiegand aster...@wiegand.name On a production system, running 1.4.17 (compiled from bristuff-0.4.0-test6-xr1) we had this strange issue two times in the last weeks: [2009-01-13 13:58:30] WARNING[1213] channel.c: Fixup failed on channel SIP/2332-081d0108MASQ, strange things may happen. [2009-01-13 13:58:30] WARNING[1213] channel.c: Hangup failed! Strange things may happen! [2009-01-13 13:58:30] WARNING[1213] channel.c: Failed to perform masquerade [2009-01-13 13:58:30] WARNING[1213] channel.c: Channel 'SIP/2332-081d0108' may not have been hung up properly and: [2009-01-23 14:27:17] WARNING[21528] channel.c: Fixup failed on channel SIP/2332-083c3778MASQ, strange things may happen. [2009-01-23 14:27:17] WARNING[21528] channel.c: Hangup failed! Strange things may happen! [2009-01-23 14:27:17] WARNING[21528] channel.c: Failed to perform masquerade [2009-01-23 14:27:17] WARNING[21528] channel.c: Channel 'SIP/2332-083c3778' may not have been hung up properly Both times all SIP channels got stuck and the CLI became inresponsive. Calls continued for a while, but new SIP calls could not be established. On the second time this happended, all SIP phones could not subscribe to the Asterisk any longer and a few minutes later the log filled with: [2009-01-23 14:43:21] ERROR[22319] chan_sip.c: Call to peer '2333' rejected due to usage limit of 10 On the CLI one could see, that there were 100s of (rejected) calls to this SIP phones. The phones that show up in the ERROR messages are in a group call made by a Dial(Local/...Local.../Local/...) construct. But other SIP phones were affected as well. It seemed like the whole chan_sip module became stuck. I also could not unload chan_sip.so, but can't remeber the exact error message it gave. The only thing that was left was to restart Asterisk. Can someone give me some clue what the 'Fixup failed ...' and 'masquerade' warnings actually mean? Any help appreciated. Udo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
[asterisk-users] Reading/Writing the Astdb
All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx database show output.txt and work with that and then set a new value such as asterisk -rx database put $key $value. The whole process can take over 1 second for EACH ENTRY which adds up for more than a few keys. What I do now is dump the entire Astdb using db_dump185 in about 0.003 ms, and then read the entire Astdb output as a hash and then manipulate key, value pairs that way. The entire process will take me less than 0.020 ms total. My problem is this. I am unable to find a corresponding way of doing this in reverse. That is, I do not have a corresponding way to write the new values back to the Astdb. The most obvious way of writing to the Astdb is by using PERL's DB_File. I tried compiling DB_File using the Berkeley 1.85 lib and header to no avail. I've had no luck with db_load either. Any insight at all to write many values to the Astdb quickly would be greatly appreciated. Regards; _ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_012009___ -- 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] Reading/Writing the Astdb
external DB? like mysql? 2009/1/24 cbbs...@hotmail.com All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx database show output.txt and work with that and then set a new value such as asterisk -rx database put $key $value. The whole process can take over 1 second for EACH ENTRY which adds up for more than a few keys. What I do now is dump the entire Astdb using db_dump185 in about 0.003 ms, and then read the entire Astdb output as a hash and then manipulate key, value pairs that way. The entire process will take me less than 0.020 ms total. My problem is this. I am unable to find a corresponding way of doing this in reverse. That is, I do not have a corresponding way to write the new values back to the Astdb. The most obvious way of writing to the Astdb is by using PERL's DB_File. I tried compiling DB_File using the Berkeley 1.85 lib and header to no avail. I've had no luck with db_load either. Any insight at all to write many values to the Astdb quickly would be greatly appreciated. Regards; -- Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out.http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_012009 ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
Olivier wrote: Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME are equipped today with PBX connected with TE/PtMP interfaces to telco BRI lines. When we address those SME, my opinion is that it's very useful to be able to support any combination of TE/NT, PtP/PtMP modes. Latest 1.6 Asterisk and 1.4.8 Libpri introduced a new set of welcomed ISDN BRI features. Unfortunately, NT/PtMP is not available at this time, in latest Zaptel/Asterisk/Libpri. My question is what is the policy concerning NT/PtMP ? Is it really hard to extend Libpri to support this mode ? Or shall mISDN remain the way to go when NT/PtMP is needed ? Hey Olivier, I actually was the one that did a lot the work in adding the BRI support to libpri/chan_dahdi. NT PTMP is very significantly different, in that you have to do much more from a TEI management perspective. Most people's needs that I saw were actually fulfilled in using either NT or TE PTP or TE PTMP, since they were interfacing with PBXs or using TE-PTMP trunks from the telephone network to provide voice trunks for Asterisk. Right now, I would not preclude the possibility that NT-PTMP support might be added, but I could not give you a concrete time at which it will be done, since it will probably require some significant internal changes in libpri. To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. Matthew Fredrickson Digium, Inc. ___ -- 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] NAT router for Linux
Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny and one important thing: I can't use a GUI to configure anything. Any help is highly apreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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] NAT router for Linux
If you set the 'bindaddr' to your private IP address, the Gtalk connection from your Asterisk server to my Gtalk client (running on Windows) works fine. That's at least what we've tested together Julien, right? If the STUN packets are properly exchanged between Asterisk and the Gtalk client you're trying to communicate with, there should not be any problem. The thing is that you and I did not test the Asterisk - NAT box - Internet - NAT -box - Gtalk client since my Gtalk client had a public IP. I don't advise you to purchase a NAT router to test this scenario though. Philippe On Sat, Jan 24, 2009 at 6:32 PM, Julien Claassen jul...@c-lab.de wrote: Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny and one important thing: I can't use a GUI to configure anything. Any help is highly apreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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 -- Philippe Sultan ___ -- 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] Logging outgoing calls
That is a good idea too, where would I configure asterisk to log the channel status on that custom field? On Sat, Jan 24, 2009 at 8:27 AM, David fire ddf...@gmail.com wrote: and what about add a custome field or setup a variable on outgoing calls and use the common cdr and then filtering by that field. David 2009/1/24 Tilghman Lesher tilgh...@mail.jeffandtilghman.com On Friday 23 January 2009 18:22:16 Pascal Bruno wrote: Is it possible to log just the outgoing calls using cdr_odbc into a custom mysql database table? my table will look like this: | call_status | |-- --| | · id | | · destination | | · status | || I just need to store the destination number and the status of the channel for example BUSY, UNAVAILABLE etc... Yes, if you install the cdr_adaptive_odbc backport. See the sample config file for more information. Web: http://svncommunity.digium.com/view/tilghman/branches/1.4 SVN: http://svncommunity.digium.com/svn/tilghman/branches/1.4 -- Tilghman ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Reading/Writing the Astdb
I second that, while read an berkeley db file outside of it's main application can work fine, writing in it would certainly lead to huge trouble (data loss, corrupted file, ...) A berkeley db file is .. a file, not a database server David fire a écrit : external DB? like mysql? 2009/1/24 cbbs...@hotmail.com mailto:cbbs...@hotmail.com All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx database show output.txt and work with that and then set a new value such as asterisk -rx database put $key $value. The whole process can take over 1 second for EACH ENTRY which adds up for more than a few keys. What I do now is dump the entire Astdb using db_dump185 in about 0.003 ms, and then read the entire Astdb output as a hash and then manipulate key, value pairs that way. The entire process will take me less than 0.020 ms total. My problem is this. I am unable to find a corresponding way of doing this in reverse. That is, I do not have a corresponding way to write the new values back to the Astdb. The most obvious way of writing to the Astdb is by using PERL's DB_File. I tried compiling DB_File using the Berkeley 1.85 lib and header to no avail. I've had no luck with db_load either. Any insight at all to write many values to the Astdb quickly would be greatly appreciated. Regards; Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_012009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.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] NAT router for Linux
Hi, Are you having problems with sip calls or just using Gtalk? If you are behind a nat router you may need to forward in to your server port 5252. Check out the /etc/asterisk/gtalk.conf and /etc/asterisk/jabber.conf files. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julien Claassen Sent: Saturday, January 24, 2009 12:32 PM To: asterisk users mailinglist Subject: [asterisk-users] NAT router for Linux Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny and one important thing: I can't use a GUI to configure anything. Any help is highly apreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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 ___ -- 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] idle-url for Cisco 7940 using Sip
Does anybody know if idle-url works for Cisco 79xx using Sip? If it doesn't work is it a Sip vs SCCP issue or Asterisk vs CallManager issue? Thanks Paul ___ -- 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] Having tone in my fxs, and loading the zaptel
Hi List; If any one faced the following problem and can help me: My zaptel version is: 1.4.10.1 My asterisk version is: 1.4.19.1 OS: Fedora core 8 I used make config for the initialization script. Now, sometimes when the hardware restarted, we discovered that no tone in the handset connected to the fxs ports. And sometimes it work normally, no rule. To resolve it, I stop asterisk and then I type modprob -r wctdm and then I restart the machine using init 6 , and sometimes it come up and somtimes I repeat this twice to come up, so what could be the issue? I know that make config is making a problem, but could it be my problem related to using make config? So I have to write the init script? If yes, I need a link to know how to write the init script manually for my digium and other cards, I do not know what is the needed lines need to be added. Also, could it be a bug related to zaptel driver so I have to download new drivers? Actually, I would like to hear if anyone face this problem and how he resolved it, so I can move based on that. Regards Bilal ___ -- 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] Newbie in Cisco Phone
I run chan_sccp at home. It works well, supports the park function, but does not make use of the conference button. I haven't used the chan_skinny, so I don't know how it compares. With chan_sccp, if you make a change to the configuration, you need to reload the module, thus taking down all phones running sccp. That's fine if there are only a couple of phones, but would be a problem if it is a big office. Mike -Original Message- From: Sam Tam [mailto:samtam...@gmail.com] Sent: Friday, January 23, 2009 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Newbie in Cisco Phone Hi I am no expert in the cisco phone Do you have time to help Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Santulli Sent: Saturday, January 24, 2009 12:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: tam...@gmail.com Subject: Re: [asterisk-users] Newbie in Cisco Phone you can try chan_sccp at www.chan-sccp.org it supports most of ccm features and all kind of cisco phones with skinny firmware. Take a look ;) If you need support you can write me back. Federico - Original Message - From: Sam Tam samtam...@gmail.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, January 23, 2009 8:56 AM Subject: Re: [asterisk-users] Newbie in Cisco Phone Well does it matter if the asterisk server is not located in the same network? I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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 -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Zaptel? Dahdi?
Is Zaptel no longer available? I returned to a long shelved project (using TDM400P and a customized, canned version of *) and, getting to the configuration, find wctdm is not there. I recall the authors where very enterprise oriented and focused on T1 cards. So they left analog support out. Anyway, before I abandon all hope and dive into the new stuff, I thought I would chase this down a bit. 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] Which policy for ISDN BRI support in NT/PtMP ?
Matthew Fredrickson wrote: [snip] I actually was the one that did a lot the work in adding the BRI support to libpri/chan_dahdi. [snip] To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. Hi Matthew, Is there a BRI status document? I'm asking because it's not clear to me if I need mISDN or that Digium (you) has developed native support for the B410P card BRI card in zaptel/dahdi/libpri. If there's native support for BRI, which version(s) of zaptel/dahdi/libpri would I need to install to test this? Thanks, Patrick ___ -- 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] NAT router for Linux
I'm using gtalk. So I can try to configure my router (it's got a lot of javascript :-) ) to forward 5222 to my server and the same thing backwards? Thanks for responding so fast! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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] NAT router for Linux
I'm not completely sure about the things my router can do. It's from the telephone company and it's supposed to do a lot of stuff. I've just heard, that windows people could solve such things. After all my setup isn't too strange or rare? Or is it for running asterisk? Kndest regards and thanks for all the testing Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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] NAT router for Linux
The short answer to your question--assuming it is the right question to be asking--is that Linux comes with a built-in NAT infrastructure as part of its packet filter (netfilter). The utility iptables is used to manage it. Simple example: echo 1 /proc/sys/net/ipv4/ip_forward iptables -t nat -A POSTROUTING -o eth0 -s 192.168.1.0/24 -j MASQUERADE Julien Claassen wrote: I'm using gtalk. So I can try to configure my router (it's got a lot of javascript :-) ) to forward 5222 to my server and the same thing backwards? Thanks for responding so fast! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] NAT router for Linux
No, your setup is not unusual for a client. If you are not happy with your router, you can set it to Ethernet bridge mode (if it's DSL over ATM transport, that's RFC1483). Then your PC behind it can hold the public Layer 3 interface, but the DSL modem will still do the ATM/G.DMT stuff. Julien Claassen wrote: I'm not completely sure about the things my router can do. It's from the telephone company and it's supposed to do a lot of stuff. I've just heard, that windows people could solve such things. After all my setup isn't too strange or rare? Or is it for running asterisk? Kndest regards and thanks for all the testing Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
Patrick wrote: Matthew Fredrickson wrote: [snip] I actually was the one that did a lot the work in adding the BRI support to libpri/chan_dahdi. [snip] To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. Hi Matthew, Is there a BRI status document? I'm asking because it's not clear to me There release logs that are made whenever we make a new release of libpri or Asterisk which contain information about development in this area. if I need mISDN or that Digium (you) has developed native support for the B410P card BRI card in zaptel/dahdi/libpri. If there's native support for BRI, which version(s) of zaptel/dahdi/libpri would I need to install to test this? You must have the most current version of DAHDI, libpri-1.4, and a version of Asterisk-1.6. Matthew Fredrickson Digium, inc. ___ -- 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] Dahdi Init script for Suse?
Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. There's practically zilch on google on the matter. I think suse support should be included by default, though. Thanks!, Joshua Kinard ___ -- 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] Zaptel? Dahdi?
On Sat, Jan 24, 2009 at 02:30:24PM -0500, j...@j4computers.com wrote: Is Zaptel no longer available? Aparantly no longer linked from asterisk.org . Still very much available from http://downloads.digium.com/pub/zaptel/ as before. I returned to a long shelved project (using TDM400P and a customized, canned version of *) and, getting to the configuration, find wctdm is not there. I recall the authors where very enterprise oriented and focused on T1 cards. So they left analog support out. Anyway, before I abandon all hope and dive into the new stuff, I thought I would chase this down a bit. If this is a new project, DAHDI is generally where you should look. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Reading/Writing the Astdb
On Sat, Jan 24, 2009 at 11:00:58AM -0500, cbbs...@hotmail.com wrote: All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx database show output.txt and work with that and then set a new value such as asterisk -rx database put $key $value. The whole process can take over 1 second for EACH ENTRY which adds up for more than a few keys. Either do that through the manager interface, or (if you want to batch commands) send them directly over the unix-domain socket asterisk.ctl . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Zaptel? Dahdi?
On 1/24/2009 at 4:20 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Jan 24, 2009 at 02:30:24PM -0500, j...@j4computers.com wrote: Is Zaptel no longer available? Aparantly no longer linked from asterisk.org . Still very much available from http://downloads.digium.com/pub/zaptel/ as before. Thanks. 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] interesting comment. New Physics?
While browsing about, found http://www.voip-info.org/wiki/view/TDM400P, where I found this comment: Here's a tip passed on from an old telephone engineer. Where your copper 2-wire cable approaches the building, underground, finish with several large loops, about a metre in diameter, laid on top of each other. Fast moving, high energy spikes will spin off the outside of the loop as they speed into your installation, reducing the amount of energy your spike trap has to absorb. The coil shouldn't have enough loops to create any induction effects. I am grateful for this introduction to the New Physics. Just be sure not to stand on the periphery of the loop, when lightning strikes nearby. g 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] interesting comment. New Physics?
For fiber installations, be sure that your loops are not placed where flashes will distract drivers or people performing potentially dangerous activities. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of j...@j4computers.com Sent: Saturday, January 24, 2009 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] interesting comment. New Physics? While browsing about, found http://www.voip-info.org/wiki/view/TDM400P, where I found this comment: Here's a tip passed on from an old telephone engineer. Where your copper 2-wire cable approaches the building, underground, finish with several large loops, about a metre in diameter, laid on top of each other. Fast moving, high energy spikes will spin off the outside of the loop as they speed into your installation, reducing the amount of energy your spike trap has to absorb. The coil shouldn't have enough loops to create any induction effects. I am grateful for this introduction to the New Physics. Just be sure not to stand on the periphery of the loop, when lightning strikes nearby. g 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] Dahdi Init script for Suse?
Hi, I've it up and running on OpenSuse 11. I used the scripts provided by the sources and commented out one line: # # Determine which kind of configuration we're using # #system=redhat # assume redhat system=debian # assume debian This forces the script to use debian style. It works for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. == INGEGNI Tech S.r.l. http://www.ingegnitech.com Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. There's practically zilch on google on the matter. I think suse support should be included by default, though. Thanks!, Joshua Kinard ___ -- 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] interesting comment. New Physics?
To be fair they did specify underground ;) j On Sat, 24 Jan 2009, Don Kelly wrote: For fiber installations, be sure that your loops are not placed where flashes will distract drivers or people performing potentially dangerous activities. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of j...@j4computers.com Sent: Saturday, January 24, 2009 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] interesting comment. New Physics? While browsing about, found http://www.voip-info.org/wiki/view/TDM400P, where I found this comment: Here's a tip passed on from an old telephone engineer. Where your copper 2-wire cable approaches the building, underground, finish with several large loops, about a metre in diameter, laid on top of each other. Fast moving, high energy spikes will spin off the outside of the loop as they speed into your installation, reducing the amount of energy your spike trap has to absorb. The coil shouldn't have enough loops to create any induction effects. I am grateful for this introduction to the New Physics. Just be sure not to stand on the periphery of the loop, when lightning strikes nearby. g 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 ___ -- 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] Passing DTMF
Hello: Yes, DTMF can be a problem on the phones themselves as Sam observed, and inband can help with this in certain situations. I have DTMF working internally in my pbx just fine though. The problem here is transmitting dtmf from my pbx through a carrier to a party who has phoned into my pbx through that carrier. I have this working on VoicePulse but only about 75% of the time. VoicePulse has indicated to me via Support that they know this is a flaw in their system and someday they plan to fix it, but no commitment about when. So my question is: Does anybody know of a carrier who handles DTMF correctly, particularly in situations where my pbx is called externally? Thanks. Chris On Sat, 24 Jan 2009, Jon Weisman wrote: since you're using ulaw try setting dtmfmode = inband if this doesnt work try = auto -Jon - Original Message - From: Christopher Gray ch...@bayareadigital.us To: Asterisk Users Listserve asterisk-users@lists.digium.com Sent: Friday, January 23, 2009 8:13 PM Subject: [asterisk-users] Passing DTMF Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the question: does anybody know of a carrier that can reliably allow an extension in my pbx to send touchtone to a calling party? I have tried Vitelity and VoicePulse. Neither can do this, and VoicePulse indicates they know it's a problem and will fix it at some unknown time in the future. For the curious, here is the reason for the need. My wife, who works as a translator, will use this extension to receive calls from companies needing translation. When she receives such a call, step 1 for her is to enter an employee id code. At the end of the call, she must enter an additional code to receive an ending time. Vitelity can't do this at all. VoicePulse works about 75% of the time which is not acceptable. Thanks for any advice. Chris Christopher Gray, President Bay Area Digital Promoting good health with innovative technology 870 Market Street, #653 San Francisco, CA 94102 Phone: (415) 217-6667 fax:(415) 962-2520 Email: ch...@bayareadigital.us ___ -- 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 Christopher Gray, President Bay Area Digital Promoting good health with innovative technology 870 Market Street, #653 San Francisco, CA 94102 Phone: (415) 217-6667 fax:(415) 962-2520 Email: ch...@bayareadigital.us ___ -- 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] interesting comment. New Physics?
Jeff LaCoursiere wrote: To be fair they did specify underground ;) j On Sat, 24 Jan 2009, Don Kelly wrote: Well sounds like the info was being passed along by someone who did not understand the purpose. I would make the loops tighter, and the point is it acts like a choke, especially in shielded cable, but you would want your network protector or ground on the outside of it, not the inside, so putting outside the building is not that great an idea. For fiber installations, be sure that your loops are not placed where flashes will distract drivers or people performing potentially dangerous activities. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of j...@j4computers.com Sent: Saturday, January 24, 2009 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] interesting comment. New Physics? While browsing about, found http://www.voip-info.org/wiki/view/TDM400P, where I found this comment: Here's a tip passed on from an old telephone engineer. Where your copper 2-wire cable approaches the building, underground, finish with several large loops, about a metre in diameter, laid on top of each other. Fast moving, high energy spikes will spin off the outside of the loop as they speed into your installation, reducing the amount of energy your spike trap has to absorb. The coil shouldn't have enough loops to create any induction effects. I am grateful for this introduction to the New Physics. Just be sure not to stand on the periphery of the loop, when lightning strikes nearby. g 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 ___ -- 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] no dial tone tdm400p
This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0 ##switchtype=national ##pridialplan=national channel=1 signalling=fxs_ks language=us context=line-1 group=0 ##switchtype=national ##pridialplan=national channel=4 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] NAT router for Linux
Thanks for the answers. I have to read those more carefully, when I'm properly awake and concentrated, but it sounds as if this might be of help. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.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] Dahdi Init script for Suse?
Stared at an init script long enough, and managed to devise up the following script. This applies straight to tools/dahdi.init in dadhi-linux-complete. Minus the top hunk in the patch (which sets system = suse), this converts it into a working script for suse systems. Thoughts? What's the likelyhood something like this could get included in an actual release? If possible, what extra work needs doing? --J From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Marco [marcota...@libero.it] Sent: Saturday, January 24, 2009 5:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Init script for Suse? Hi, I've it up and running on OpenSuse 11. I used the scripts provided by the sources and commented out one line: # # Determine which kind of configuration we're using # #system=redhat # assume redhat system=debian # assume debian This forces the script to use debian style. It works for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. == INGEGNI Tech S.r.l. http://www.ingegnitech.com Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. There's practically zilch on google on the matter. I think suse support should be included by default, though. Thanks!, Joshua Kinard ___ -- 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 dahdi.init-suse.patch Description: dahdi.init-suse.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] Dahdi Init script for Suse?
On Sat, 2009-01-24 at 23:45 +0100, Marco wrote: Hi, I've it up and running on OpenSuse 11. I used the scripts provided by the sources and commented out one line: # # Determine which kind of configuration we're using # #system=redhat # assume redhat system=debian # assume debian This forces the script to use debian style. It works for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. Just wondering... The O.P. said he's using SLE. You're talking about openSUSE. Are you using the rpm's from the OBS? The zaptel-rpm for 10.3 were containing the proper startup scripts. I've got some suse machines running asterisk, but as soon as hw get's involved, i'm stuck: neither pri, nor bri (mISDN) seems to be working on anything later than 10.3. Hans ___ -- 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] monitoring SIP connection
with dahdi I can monitor hardware cards with dahdi show status. I can then tell if a T1/PRI card goes into condition RED. When I have a VOIP/SIP connection to lets say Call Manager how can I monitor this connection? Today I suddenly started getting 503 service not available messages when trying to use CCM to place a call. It would have been nice to know ahead of time when something changed and I could no longer make calls. sip show peers did not report anything wrong either. How can I monitor a SIP connection? Jerry ___ -- 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] monitoring SIP connection
Jerry Geis schrieb:
with dahdi I can monitor hardware cards with dahdi show status.
I can then tell if a T1/PRI card goes into condition RED.
When I have a VOIP/SIP connection to lets say Call Manager
how can I monitor this connection?
Today I suddenly started getting 503 service not available messages
when trying to use CCM to place a call.
It would have been nice to know ahead of time when something changed
and I could no longer make calls.
sip show peers did not report anything wrong either.
sip show registry maybe?
How can I monitor a SIP connection?
Use asterisk or sipsak or somesuch to place a call to a test
extension every minute?
test = {
Answer();
Hangup();
}
Philipp Kempgen
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[asterisk-users] asterisk help
hello! i'm new to asterisk. i'm using CentOS 5.2 + ASterisk 1.6 when i finish installing asterisk, i configure sip.conf like: [4455] type=friend username=4455 secret=1234 host=dynamic context=internal [4466] type=friend username=4466 secret=1234 host=dynamic context=internal and extensions.conf like: [internal] exten = 4455,1,Dial(SIP/4455) exten = 4466,1,Dial(SIP/4466) ok. i start asterisk with: #asterisk -cvvv and open a softphone try to connect and nothing! i try nmap @ port 5060 but it's closed! what i can do? thx _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- 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] Choppy Sound On Bridging From SIP-IAX
I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX-IAX peers or SIP-SIP peers. My timing source is ztdummy. Does anybody have any ideas on the possible source of the problem? -- Muiz Motani m...@askaritech.com Askari Technologies ___ -- 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] Passing DTMF
Christopher, did you receive the email that I sent to your yesterday? It was delivered Jan 23 20:47:31 -0600. Maybe it went to your junk box.. I will try again. Sam Christopher Gray wrote: Hello: Yes, DTMF can be a problem on the phones themselves as Sam observed, and inband can help with this in certain situations. I have DTMF working internally in my pbx just fine though. The problem here is transmitting dtmf from my pbx through a carrier to a party who has phoned into my pbx through that carrier. I have this working on VoicePulse but only about 75% of the time. VoicePulse has indicated to me via Support that they know this is a flaw in their system and someday they plan to fix it, but no commitment about when. So my question is: Does anybody know of a carrier who handles DTMF correctly, particularly in situations where my pbx is called externally? Thanks. Chris On Sat, 24 Jan 2009, Jon Weisman wrote: since you're using ulaw try setting dtmfmode = inband if this doesnt work try = auto -Jon - Original Message - From: Christopher Gray ch...@bayareadigital.us To: Asterisk Users Listserve asterisk-users@lists.digium.com Sent: Friday, January 23, 2009 8:13 PM Subject: [asterisk-users] Passing DTMF Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the question: does anybody know of a carrier that can reliably allow an extension in my pbx to send touchtone to a calling party? I have tried Vitelity and VoicePulse. Neither can do this, and VoicePulse indicates they know it's a problem and will fix it at some unknown time in the future. For the curious, here is the reason for the need. My wife, who works as a translator, will use this extension to receive calls from companies needing translation. When she receives such a call, step 1 for her is to enter an employee id code. At the end of the call, she must enter an additional code to receive an ending time. Vitelity can't do this at all. VoicePulse works about 75% of the time which is not acceptable. Thanks for any advice. Chris Christopher Gray, President Bay Area Digital Promoting good health with innovative technology 870 Market Street, #653 San Francisco, CA 94102 Phone: (415) 217-6667 fax:(415) 962-2520 Email: ch...@bayareadigital.us ___ -- 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 Christopher Gray, President Bay Area Digital Promoting good health with innovative technology 870 Market Street, #653 San Francisco, CA 94102 Phone: (415) 217-6667 fax:(415) 962-2520 Email: ch...@bayareadigital.us ___ -- 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] Choppy Sound On Bridging From SIP-IAX
Muiz Motani wrote: I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX-IAX peers or SIP-SIP peers. My timing source is ztdummy. Does anybody have any ideas on the possible source of the problem? All of my chopping problems where because of firewalls and a few of them I never figured out but they magically started working right. Can you check any firewall logs for dropped packets? ___ -- 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 do a transfer in agi
Hey,
I am trying to work through a use case requirement where a user
listens to a some advertisement and then if at the end off it they
press a key they press a 1 key they get transfered to a pre-defined
number. I am using the asterisk java library at http://asterisk-java.org/
.
I was able to originate the call easily by doing a
String channel=SIP/ + phoneNumber + @+getSipPeer();
this.getAsteriskServer().originateToExtension(channel,
getContext(), getExtension(), getPriority(), getTimeout(),
getCallerId(), vars);
my extensions.conf file has this defined in it
exten = 9001,1,Agi(agi://127.0.0.1:4573/${campaign}.agi?campaign=$
{campaign})
A fast AGI script is then called and I can play the media.
My first option was at the end of the script to set the extension to
continue at 9002
channel.setContext(davidtest);
channel.setExtension(9002);
channel.setPriority(1);
exten = 9002,1,Transfer(SIP/${campaignnumb...@ser1)
When I tried this I got the following in the logs and the call would
be dropped.
[Jan 8 23:33:35] VERBOSE[16163] logger.c: --- (8 headers 0 lines)
---
[Jan 8 23:33:39] VERBOSE[16163] logger.c:
--- SIP read from 10.128.181.23:5060 ---
SIP/2.0 403 Forbidden^M
Via: SIP/2.0/UDP
209.34.91.75
:5060;received=10.128.181.21;branch=z9hG4bK6937f2b6;rport=5060^M
From: companyname sip:+1415...@209.34.91.75;tag=as1e339e25^M
To: sip:415...@209.34.91.74;tag=gK028a94eb^M
Call-ID: 4b07e5ef5f04f5dc40ce7d6b4b002...@209.34.91.75^m
CSeq: 103 REFER^M
Content-Length: 0^Mites
I did try to execute the Transfer command directly in the Fast AGI
script. The call would not drop but it did not connect. I saw no
errors in the logs.
int reply = channel.exec(Transfer, SIP/415...@ser1);
I am new to Asterisk and VOIP so any help would be appreciated,
cheers,
David
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Re: [asterisk-users] monitoring SIP connection
Jerry Geis wrote: with dahdi I can monitor hardware cards with dahdi show status. I can then tell if a T1/PRI card goes into condition RED. When I have a VOIP/SIP connection to lets say Call Manager how can I monitor this connection? Today I suddenly started getting 503 service not available messages when trying to use CCM to place a call. It would have been nice to know ahead of time when something changed and I could no longer make calls. sip show peers did not report anything wrong either. How can I monitor a SIP connection? Jerry At the risk of asking a dumb question, are you already using 'qualify=yes' for the sip peers you want to monitor? -- Paul ___ -- 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] no dial tone tdm400p
On Sat, Jan 24, 2009 at 06:38:58PM -0500, j...@j4computers.com wrote: This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. What version of asterisk is it? What is the output of: cat /proc/zaptel/* asterisk -rx 'zap show channels' zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0 ##switchtype=national ##pridialplan=national Use ';' for comments. '#' are for preprocessor directives. This happens to also disable almost any line, but also create an annoying warning in the logs. Have you looked at the logs, BTW? channel=1 This only defines the first port . What about 2? signalling=fxs_ks language=us context=line-1 group=0 ##switchtype=national ##pridialplan=national channel=4 Same comments: what about 3? It is also normally kind of pointless to put both phones and POTS lines in the same group (group 0, in your case). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
