[Asterisk-Users] iaxphone for ubuntu 5.10
Hi, does anybody know if there's a iax phone running on ubuntu 5.10 which can be used with asterisk? Seems like kiax has got many compiling and libraries problems. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxphone for ubuntu 5.10
Google is your friend, from http://www.voip-info.org/wiki-VOIP+Phones : http://iaxclient.sourceforge.net/iaxcomm/index.html http://www.asteriskguru.com/tools/idefisk_beta.php http://www.voip-info.org/wiki/view/GnoPhone Giorgio Incantalupo wrote: Hi, does anybody know if there's a iax phone running on ubuntu 5.10 which can be used with asterisk? Seems like kiax has got many compiling and libraries problems. TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXPhone
Hi, Im looking for IAXphone 2.0 (from sokol-associates) source code but the site is unavailable did some one can help me please. ?? thanks _ Erwan Desvergnes - ANDIUM - 82/86 rue Château Gaillard 69100 Villeurbanne Tel. 04 3743 44 45 / Fax 04 37 43 44 44 E-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] IAXphone - ip address - extension number.
Yeee-h! doesn't this look pretty? *** Asterisk Ready. *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, IAX2/z2|20|tr) in new stack -- Called z2 -- Call accepted by 192.168.0.202 (format ulaw) -- Format for call is ulaw -- IAX2/z2/2 is ringing -- IAX2/z2/2 answered IAX2/[EMAIL PROTECTED]/1 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/1 and IAX2/z2/2 -- Channel 'IAX2/[EMAIL PROTECTED]/1' ready to transfer -- Channel 'IAX2/z2/2' ready to transfer -- Releasing IAX2/z2/2 and IAX2/[EMAIL PROTECTED]/1 -- Hungup 'IAX2/z2/2' == Spawn extension (geograph, 202, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/1' -- Hungup 'IAX2/[EMAIL PROTECTED]/1' -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 *CLI *** Rich! Carlos! - Pizzas on me when you come to Cape Town. (or I get to where you are - where's that?) What a learning curve - big thanks and let me give you a suggestion: Take leave the week after next - I'm going to be plugging in 2 internal ISDN BRI cards ;-) (next week will be to sort out the choppy sound to move from my SuSE 9.3 toy box to the production FC3 (dare I try FC4) box - so maybe take next week off as well :-) (Carlos - I'll respond to your zaprtc query later today) Cheers thanks - sincerely hope to be able to return the effort one day. regards to all, Zoltan Rich Adamson wrote: all0w=ulaw all0w=alaw all0w=gsm Look closely at the above four lines. In the allow statement, that appears to be a zero. Change that to allow. Also, I don't know which codecs the phone supports, but you might start playing with disallow=all allow=ulaw and go from there. you're 100% right - I saw the typo when the lines were commented out and the codecs were in the [z1] section. I then changed back in order to shorten the iax.conf file but forgot about the typos. Thanks - it could've taken many more hours for me to notice them again :-) [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 If you look at /usr/src/asterisk/configs/iax.conf.sample, you'll find that dtmfmode=rfx2833 is not a valid iax statement. Plus its spelled wrong (its rfc2833). Remove it, but add it into your sip.conf if you're going to play with sip. jeeze - dislexia rulz (never change a config file when in a hurry to do something else) *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 Here is the key: That is telling you it can't find a compatible codec to allow the call to complete. That's the basis for the comments above about the allow=ulaw. *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Note the above IAX. I think that should be IAX2, so look in your extensions.conf for a dial statement that looks like Dial(IAX/ and change it to Dial(IAX2/. Yep - this too would have taken me a while to notice. ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: Zoltan, If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. I only tried once to make zaprtc - this was whilst ztdummy was not loading for me. I'll go through what I did in a later email. For now, the ztdummy thing: FC2 is 2.4 and not 2.6 kernel. I remember seeing somewhere that ztdummy reacts better on 2.6 kernels. (is this true anyone???) The thing that killed my ztdummy was that, at make time, I did not have/see/notice and udev errors (so although having read it, I ignored the README.udev file), however whenever I modprobed zaptel before modprobe ztdummy (this is the order it must be done in), it would not load. gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf :-( line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # Tzafrir Cohen picked up this error on /dev and pointed me to /usr/src/zaptel-1.0.9/README.udev and once I had dumped those 6 lines into /etc/udev/rules.d/50-udev.rules the modprobe sequence worked. You need to find out if FC2 uses udev or not. (I'll re-run my attempts at zaprtc sort out the emails I got - and email you again just now) Cheers, Zoltan Regarding Ethereal, you should do tcpdump -I ethx (x=number of the port) -W filename, when you finish close it with Ctrl-z, and then you can see the file on the Asterisk or move it to another computer with Etherreal and open it (That is the way I do, so I see what Asterisk gets). Have a great weekend. Carlos ___ 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] IAXphone - ip address - extension number.
What a learning curve - big thanks and let me give you a suggestion: Take leave the week after next - I'm going to be plugging in 2 internal ISDN BRI cards ;-) I have never played with a BRI, so won't be able to help on that one. But, there are plenty of folks on the list that have it working. (next week will be to sort out the choppy sound to move from my SuSE 9.3 toy box to the production FC3 (dare I try FC4) box - so maybe take next week off as well :-) I'm using FC3 and it works well. I'd stay away from FC4 for now simply because there are not very many people on this list that have tried it, so help is likely to be almost non-existant. If you try FC3, be sure to read the READMEs in zaptel source directory, etc. There are some additional things you will need to do and it seems a lot of folks miss reading those items. In particular, look for udev stuff. Rich ___ 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] IAXphone - ip address - extension number.
Hi Carlos, OK - I speak from memory and a little bit of newbie fiddling (which thanks to you and Rich took a successful turn). Carlos Alperin wrote: Zoltan, If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I was under the impression that timing signals can be gotten from digium cards, uhci_usb (using ztdummy) and the rtc (using zaptelrtc). If you follow the thread from http://lists.digium.com/pipermail/asterisk-users/2004-September/063355.html then it suggests that from 2.6 kernel, ztdummy no longer requires USB . zaptelrtc does however require that rtc is *not* built into your kernel so you would have to recompile it without rtc if it is. For me the only ztdummy issue was with udev (see README.udev in the zaptel-1.0.9 folder) and all you would have to do was to check if FC2 has udev or not. Dont forget to modprobe zaptel before modprobe ztdummy before loading asterisk. HTH, Zoltan I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. Regarding Ethereal, you should do tcpdump -I ethx (x=number of the port) -W filename, when you finish close it with Ctrl-z, and then you can see the file on the Asterisk or move it to another computer with Etherreal and open it (That is the way I do, so I see what Asterisk gets). Have a great weekend. Carlos ___ 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] IAXphone - ip address - extension number.
If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Gotta bounce - chat tomorrow. Cheers, Zoltan ___ 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] IAXphone - ip address - extension number.
If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Okay, just fired up the laptop and it registered with our office system just fine using iax2. The laptop is RHv9 with cvs-head from 4/25/05 with no modifications of any sort. The lsmod lists zaptel only, Used = 0, and no ztdummy listed. ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Okay, just fired up the laptop and it registered with our office system just fine using iax2. The laptop is RHv9 with cvs-head from 4/25/05 with no modifications of any sort. The lsmod lists zaptel only, Used = 0, and no ztdummy listed. ___ 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 Hmmm - interesting. I did an rmmod on both ztdummy and zaptel and then fired up asterisk. I could still make a call from 1 extension to another, but both music_on_hold *and* IAX2 complained about timing (but I could still make the call!!). I haven't played with MOH yet, so I dont know if the sound will work or be choppy. Let the mystery remain? Cheers, Zoltan. *** [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Jul 10 19:20:53 WARNING[3548]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' *** cut a bit out ** [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Jul 10 19:20:53 WARNING[3548]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such file or directory == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found -- Seeding 'z1' at 192.168.0.201:4569 for 60 -- Seeding 'z2' at 192.168.0.202:4569 for 60 == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' *** ___ 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] IAXphone - ip address - extension number.
Ok, It will sound stupid, but then the process is zlib first, and asterisk after, and that is all? Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, July 10, 2005 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). ___ 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] IAXphone - ip address - extension number.
If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. FWIW, I've been running asterisk on a RHv9 laptop (for client demo purposes), connecting it via iax to our office system, and never worried about ztdummy, etc. Obviously, the laptop has no zap cards. This demo never includes meetme, etc. I'm not actually surprised. In the Makefile you'll see somewhere around line 61 that ztdummy is commented out and, if you *want* ztdummy, you should remove the # before running make linux26. However, if you follow through the use of the variable names, you will see that ztdummy is suffiex anyway so whether you comment it in or out, ztdummy gets compiled. It would me interesting to do an lsmod | grep z on your laptop to see if zaptel ztdummy are loaded. I also have a backup system (to our office system) running RHv9, and it connects and functions just fine to the primary office system via iax. Sip phones work fine on this backup system as well. That backup system was just recently replaced with an FC3 system, and I have no doubt whatsoever it will function just fine without zap cards although that system has not yet been configured or tested. So, don't be too concerned with making ztdummy (etc) function unless you truly want to support those asterisk apps that need it (eg, meetme and whatever else the wiki points out). (I seem to remember IAX needed timing, which is why I was on the ztdummy mission - no cards in my testbox either.) Okay, just fired up the laptop and it registered with our office system just fine using iax2. The laptop is RHv9 with cvs-head from 4/25/05 with no modifications of any sort. The lsmod lists zaptel only, Used = 0, and no ztdummy listed. Hmmm - interesting. I did an rmmod on both ztdummy and zaptel and then fired up asterisk. I could still make a call from 1 extension to another, but both music_on_hold *and* IAX2 complained about timing (but I could still make the call!!). I haven't played with MOH yet, so I dont know if the sound will work or be choppy. It's been awhile, but MOH and Meetme are two apps/functions that do require zap timing. There might be other apps as well, I just don't recall which (if any) but they are likely listed on the wiki. ___ 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] IAXphone - ip address - extension number.
Tzafrir Cohen wrote: On Fri, Jul 08, 2005 at 11:12:37PM +0200, Zoltan Szecsei wrote: Is this how the modprobes are supposed to respond?? gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # If you use 2.4, consider 2.6, as its ztdummy works better. If you use 2.6, you may be using udev, and need to read README.udev . Bingo! I had read the README.udev, but had not noticed any make-time udev related messages so chose to ignore its contents. Bad, bad boy - naughty me. Anyway, dumping those lines into the 50-udev-rules file has solved this issue perfectly. I can now modprobe zaptel then ztdummy with no errors and asterisk loads opens all the IAX stuff confirms that it is listening on port 4569. Cool - well spotted. Now to get rid of these darn issues: *CLI Jul 9 10:23:46 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! -- Accepting unauthenticated call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, IAX/z2|20|tr) in new stack Jul 9 10:23:57 WARNING[8102]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 10:23:57 NOTICE[8102]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 10:24:07 WARNING[8102]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/5' Jul 9 10:24:18 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:24:20 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 I'm sure its now possibly a phone setup, but being green at this asterisk stuff, I let them sell me the phones without setup manuals or anything. grrr. In response to the ethereal queries I was asked: All PCs phones on this network plug into the same 3com switch, port 8 of this switch links to a netgear DG834 (my ADSL modem/router/firewall). The three remaining ports on the netgear are unused, but if I plug anything into them, the netgear DHCP offers them a compatible IP and they can see everything plugged into the 3COM. Note that *all* equipment that is permanently connected has a *fixed* IP and does not get one from the netgear DHCP service. In short, the fact that ethereal saw nothing coming from the phones suggest a phone setup issue. However the fact that when I dial *1202 (*1 being the phones IAX dial-prefix), asterisk responds to the request, shows that the phones do however communicate in some recognisable way. Maybe I should also fiddle with a minimum iax.conf and extensions.conf. Chat soon thanks, Zoltan ___ 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] IAXphone - ip address - extension number.
If you use 2.4, consider 2.6, as its ztdummy works better. If you use 2.6, you may be using udev, and need to read README.udev . Bingo! I had read the README.udev, but had not noticed any make-time udev related messages so chose to ignore its contents. Bad, bad boy - naughty me. Anyway, dumping those lines into the 50-udev-rules file has solved this issue perfectly. I can now modprobe zaptel then ztdummy with no errors and asterisk loads opens all the IAX stuff confirms that it is listening on port 4569. Cool - well spotted. Now to get rid of these darn issues: *CLI Jul 9 10:23:46 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 9 10:23:48 NOTICE[8102]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! -- Accepting unauthenticated call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, IAX/z2|20|tr) in new stack Jul 9 10:23:57 WARNING[8102]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 10:23:57 NOTICE[8102]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 10:24:07 WARNING[8102]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/5' Jul 9 10:24:18 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 9 10:24:20 NOTICE[8102]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 I'm sure its now possibly a phone setup, but being green at this asterisk stuff, I let them sell me the phones without setup manuals or anything. grrr. In response to the ethereal queries I was asked: All PCs phones on this network plug into the same 3com switch, port 8 of this switch links to a netgear DG834 (my ADSL modem/router/firewall). The three remaining ports on the netgear are unused, but if I plug anything into them, the netgear DHCP offers them a compatible IP and they can see everything plugged into the 3COM. Note that *all* equipment that is permanently connected has a *fixed* IP and does not get one from the netgear DHCP service. In short, the fact that ethereal saw nothing coming from the phones suggest a phone setup issue. However the fact that when I dial *1202 (*1 being the phones IAX dial-prefix), asterisk responds to the request, shows that the phones do however communicate in some recognisable way. Maybe I should also fiddle with a minimum iax.conf and extensions.conf. Now we're getting there. In one of your previous emails, you indicated: 8) IAX username - still left blank 9) IAX password - still left blank Edit those to something valid, don't leave them blank. Then, in iax.conf, enter the same username and password. username= secret= Power cycle the phone and _now_ it should register properly. (That's why you are getting register_verify: Empty registration from 192.168.0.202 in the above CLI. To jump ahead a little, if the phone has a config entry for type of dtmf, set it to rfc2833. This applies more to the sip use then it does to iax, but set it anyway. Also, if the phone's config has a parameter that says something about transmit silence (or words that are something close), be sure to set that to yes. Regarding your comments about the 3com and netgear switches, ethernet switches do not forward all packets to every port. They are smart enough to know where each MAC resides, and only forward packets out a switch port if the packet is destined for the device attached to that port. So, in your ethereal packet traces all you will ever see is broadcast packets (which are sent out all ports). If you need to run ethereal again with those switches, you will have to install and run it on the asterisk box. Otherwise, you will never see the desired traffic. Rich ___ 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] IAXphone - ip address - extension number.
Ha! yes - we are getting there - hopefully soon you will allow yourself some time for anything other than me. see inbetween - and then at the end. Rich Adamson wrote: Now we're getting there. In one of your previous emails, you indicated: 8) IAX username - still left blank 9) IAX password - still left blank Edit those to something valid, don't leave them blank. Then, in iax.conf, enter the same username and password. username= secret= did that earlier today and started getting different errors, so I knew I was on the right track. Power cycle the phone and _now_ it should register properly. (That's why you are getting register_verify: Empty registration from 192.168.0.202 in the above CLI. yep To jump ahead a little, if the phone has a config entry for type of dtmf, set it to rfc2833. This applies more to the sip use then it does to iax, but set it anyway. didnt see it anywhere on the phone web-setup, but I've set this in iax.conf Also, if the phone's config has a parameter that says something about transmit silence (or words that are something close), be sure to set that to yes. couldn't find anything like this Regarding your comments about the 3com and netgear switches, ethernet switches do not forward all packets to every port. They are smart enough to know where each MAC resides, and only forward packets out a switch port if the packet is destined for the device attached to that port. So, in your ethereal packet traces all you will ever see is broadcast packets (which are sent out all ports). If you need to run ethereal again with those switches, you will have to install and run it on the asterisk box. Otherwise, you will never see the desired traffic. I was running ethereal on the same box as asterisk!!! Rich OK, Based on last nights breakthru this mornings fiddling, I have minimised iax.conf filled in everything on the phone itself. Hallelujah! (I'm sure Rich Carlos will agree) :-) I'm still not ringing the other phone, but that is now surely a dialplan issue - extensions.conf has been totally ignored and that can be tomorrows fun as my wife I have a nice dinner date tonight. * iax.conf: *** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 all0w=ulaw all0w=alaw all0w=gsm jitterbuffer=yes [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 [z2] type=friend user=z2 secret=z2 context=geograph host=dynamic dtmfmode=rfx2833 *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Jul 9 16:49:59 WARNING[13788]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 16:49:59 NOTICE[13788]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 16:50:09 WARNING[13788]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/3' -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Accepting AUTHENTICATED call from 192.168.0.202, requested format = 4, actual format = 256 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, IAX/z1|20|tr) in new stack Jul 9 16:51:29 WARNING[13788]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 16:51:29 NOTICE[13788]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 16:51:39 WARNING[13788]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/1' *CLI stop now ___ 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] IAXphone - ip address - extension number.
Based on last nights breakthru this mornings fiddling, I have minimised iax.conf filled in everything on the phone itself. Hallelujah! (I'm sure Rich Carlos will agree) :-) I'm still not ringing the other phone, but that is now surely a dialplan issue - extensions.conf has been totally ignored and that can be tomorrows fun as my wife I have a nice dinner date tonight. * iax.conf: *** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 all0w=ulaw all0w=alaw all0w=gsm Look closely at the above four lines. In the allow statement, that appears to be a zero. Change that to allow. Also, I don't know which codecs the phone supports, but you might start playing with disallow=all allow=ulaw and go from there. [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 If you look at /usr/src/asterisk/configs/iax.conf.sample, you'll find that dtmfmode=rfx2833 is not a valid iax statement. Plus its spelled wrong (its rfc2833). Remove it, but add it into your sip.conf if you're going to play with sip. *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 Here is the key: That is telling you it can't find a compatible codec to allow the call to complete. That's the basis for the comments above about the allow=ulaw. *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Note the above IAX. I think that should be IAX2, so look in your extensions.conf for a dial statement that looks like Dial(IAX/ and change it to Dial(IAX2/. ___ 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] IAXphone - ip address - extension number.
Zoltan, If you don't mind, can you explain me a little more the ztcfg problem. My experience with that is null, but I need to setup a box for testing purposes only, and I don't want to but a TDM card only for make it work. I know people that has asterisk running on SIP that don't use zaptel Zapata just they do sip also to their provider (which is not my my case) but I never did that. So I plan to install it with Ztdummy on top of FC2. Regarding Ethereal, you should do tcpdump -I ethx (x=number of the port) -W filename, when you finish close it with Ctrl-z, and then you can see the file on the Asterisk or move it to another computer with Etherreal and open it (That is the way I do, so I see what Asterisk gets). Have a great weekend. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Saturday, July 09, 2005 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Ha! yes - we are getting there - hopefully soon you will allow yourself some time for anything other than me. see inbetween - and then at the end. Rich Adamson wrote: Now we're getting there. In one of your previous emails, you indicated: 8) IAX username - still left blank 9) IAX password - still left blank Edit those to something valid, don't leave them blank. Then, in iax.conf, enter the same username and password. username= secret= did that earlier today and started getting different errors, so I knew I was on the right track. Power cycle the phone and _now_ it should register properly. (That's why you are getting register_verify: Empty registration from 192.168.0.202 in the above CLI. yep To jump ahead a little, if the phone has a config entry for type of dtmf, set it to rfc2833. This applies more to the sip use then it does to iax, but set it anyway. didnt see it anywhere on the phone web-setup, but I've set this in iax.conf Also, if the phone's config has a parameter that says something about transmit silence (or words that are something close), be sure to set that to yes. couldn't find anything like this Regarding your comments about the 3com and netgear switches, ethernet switches do not forward all packets to every port. They are smart enough to know where each MAC resides, and only forward packets out a switch port if the packet is destined for the device attached to that port. So, in your ethereal packet traces all you will ever see is broadcast packets (which are sent out all ports). If you need to run ethereal again with those switches, you will have to install and run it on the asterisk box. Otherwise, you will never see the desired traffic. I was running ethereal on the same box as asterisk!!! Rich OK, Based on last nights breakthru this mornings fiddling, I have minimised iax.conf filled in everything on the phone itself. Hallelujah! (I'm sure Rich Carlos will agree) :-) I'm still not ringing the other phone, but that is now surely a dialplan issue - extensions.conf has been totally ignored and that can be tomorrows fun as my wife I have a nice dinner date tonight. * iax.conf: *** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 all0w=ulaw all0w=alaw all0w=gsm jitterbuffer=yes [z1] type=friend user=z1 secret=z1 context=geograph host=dynamic dtmfmode=rfx2833 [z2] type=friend user=z2 secret=z2 context=geograph host=dynamic dtmfmode=rfx2833 *** asterisks response as I dial Asterisk Ready. *CLI iax2 show p peers provisioning *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (D) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (D) 255.255.255.255 4569 Unmonitored *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 z2003 geograph No z1 z1003 geograph No *CLI -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 256 *-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX/z2|20|tr) in new stack* Jul 9 16:49:59 WARNING[13788]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 9 16:49:59 NOTICE[13788]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 9 16:50:09 WARNING[13788]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/3' -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569 -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569 -- Accepting AUTHENTICATED call from 192.168.0.202
RE: [Asterisk-Users] IAXphone - ip address - extension number.
Ok, Now you get registration, but still you cannot complete the call. I don't see your dialplan on your files. However, the error log shows that ztdummy is not working, so there is no timming. Lsmod shows the module already installed? Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 12:10 PM To: Time Bandit; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Time Bandit wrote: *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. OK, my iax.conf now has: [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=192.168.0.201 mailbox=1201 context=geograph callerid=Zoltan1 201 [z2] type=friend host=192.168.0.202 mailbox=1202 context=geograph callerid=Zoltan2 202 **Then I load asterisk -vvvc: Asterisk Ready. *CLI *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (S) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (S) 255.255.255.255 4569 Unmonitored *CLI *CLI *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 -no secret- 003 geograph No z1 -no secret- 003 geograph No *CLI *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). gl0:/home/zls # ** Dialing from one phone ** on phone z1 I go offhook (hear dialtone); dial 202 and get enagaged tone. Similar if I dial 1, or 2, or 201, or 202 or anything in fact. If I go through the menu on the handset, these are the correct IP addresses of the phones. I can ping these phone IP addresses from any other PC on the network. *My extensions.conf is:*** [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [geograph] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() exten = 201,1,Dial(IAX/z1,20,tr) exten = 202,1,Dial(IAX/z2,20,tr) exten = 1201,1,Dial(IAX/z1,20,tr) exten = 1202,1,Dial(IAX/z2,20,tr) The last 4 entries in /var/log/asterisk/messages is: * Jul 8 17:57:32 WARNING[5219]: Unable to open pseudo channel for timing... Sound may be choppy. Jul 8 17:57:32 WARNING[5219]: Unable to open IAX timing interface: No such file or directory Jul 8 17:57:32 NOTICE[5219]: Unable to load config sip.conf, SIP disabled Jul 8 17:57:32 WARNING[5219]: Unable to get our IP address, Skinny disabled Jul 8 17:57:32 WARNING[5219]: Read error on sound device: Resource temporarily unavailable My head is: Bald please help. TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) In my reply to time bandits suggestion, that is exactly what I did and iax2 show peers now shows the IP address. *but* I still cannot dial anywhere. What you have insinuated in your comment is that Asterisk and the phone should have automatically sorted themselves out and because this is not happening, my problem is elsewhere - proven by the fact that I have hardcoded host=192.168.0.201 in the iax.conf file (whereas before I used to have host=dynamic) and it still does not work. Possibilities: 1) The iax timing error when I load asterisk (see reply to time bandit) 2) The phone itself. They are 2 new Advantage Century Telecomms Corp AC-104ALDU units which I bought because they were both SIP and IAX compatible. No documentation came with them and on the web page I could only find SLD specs, not model ALDU specs. Hm - I wonder how good that IAX claim is. http://www.act-tel.com.tw/_UpLoad/Pic/P104S-Spec.pdf Any suggestions how I should proceed? TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
Ha! I think I found my phone - see point 2 on my previous post (copied below). Although the sticker on the bottom of my phone does have ALDU, the firmware version is 02.09.07 So, it should cope with IAX only (and no SIP at all), yet I cannot get it registered (yet) http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone Sorry to add to my prev post/request. regards, Zoltan Rich Adamson wrote: The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) In my reply to time bandits suggestion, that is exactly what I did and iax2 show peers now shows the IP address. *but* I still cannot dial anywhere. What you have insinuated in your comment is that Asterisk and the phone should have automatically sorted themselves out and because this is not happening, my problem is elsewhere - proven by the fact that I have hardcoded host=192.168.0.201 in the iax.conf file (whereas before I used to have host=dynamic) and it still does not work. Possibilities: 1) The iax timing error when I load asterisk (see reply to time bandit) 2) The phone itself. They are 2 new Advantage Century Telecomms Corp AC-104ALDU units which I bought because they were both SIP and IAX compatible. No documentation came with them and on the web page I could only find SLD specs, not model ALDU specs. Hm - I wonder how good that IAX claim is. http://www.act-tel.com.tw/_UpLoad/Pic/P104S-Spec.pdf Any suggestions how I should proceed? TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: Ok, Now you get registration, but still you cannot complete the call. I don't see your dialplan on your files. However, the error log shows that ztdummy is not working, so there is no timming. Lsmod shows the module already installed? Carlos I did have it on a previous post, but here again: (note I have no special HW so I did not modprobe zapte before modprobe ztdummy - not sure if that is relevant or not) gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # * extensions.conf * [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [geograph] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() exten = 201,1,Dial(IAX/z1,20,tr) exten = 202,1,Dial(IAX/z2,20,tr) exten = 1201,1,Dial(IAX/z1,20,tr) exten = 1202,1,Dial(IAX/z2,20,tr) *** /var/log/messages ** Jul 8 19:40:03 WARNING[6707]: Unable to open pseudo channel for timing... Sound may be choppy. Jul 8 19:40:03 WARNING[6707]: Unable to open IAX timing interface: No such file or directory Jul 8 19:40:03 NOTICE[6707]: Unable to load config sip.conf, SIP disabled Jul 8 19:40:03 WARNING[6707]: Read error on sound device: Resource temporarily unavailable Jul 8 19:40:03 WARNING[6707]: Unable to get our IP address, Skinny disabled *** ___ 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] IAXphone - ip address - extension number.
Ha! (again) Hopefully this helps you guys a bit (to help me ) Ta yet again, Zoltan From the wiki page about my phone, I changed my iax.conf to: (note 6 lines after CallerID) * sip.conf ** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=192.168.0.201 mailbox=1201 context=geograph callerid=Zoltan1 201 qualify=yes disallow=all allow=ulaw allow=alaw allow=g729 dtmfmode=info [z2] type=friend host=192.168.0.202 mailbox=1202 context=geograph callerid=Zoltan2 202 qualify=yes disallow=all allow=ulaw allow=alaw allow=g729 dtmfmode=info ** end sip.conf asterisk -vvvc *** Asterisk Ready. *CLI Jul 8 19:46:21 NOTICE[6826]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 8 19:46:21 NOTICE[6826]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (S) 255.255.255.255 4569 UNREACHABLE z1 192.168.0.201 (S) 255.255.255.255 4569 UNREACHABLE *CLI ___ 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] IAXphone - ip address - extension number.
If I write try with sip, everybody will try to hang me in the highest tree. So, If was on your position, I'll try SIP. I'm trying to help, no to argue with anyone, and of course not to dissinform or missinterpretate anything. Just help. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Rich Adamson wrote: The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) In my reply to time bandits suggestion, that is exactly what I did and iax2 show peers now shows the IP address. *but* I still cannot dial anywhere. What you have insinuated in your comment is that Asterisk and the phone should have automatically sorted themselves out and because this is not happening, my problem is elsewhere - proven by the fact that I have hardcoded host=192.168.0.201 in the iax.conf file (whereas before I used to have host=dynamic) and it still does not work. Possibilities: 1) The iax timing error when I load asterisk (see reply to time bandit) 2) The phone itself. They are 2 new Advantage Century Telecomms Corp AC-104ALDU units which I bought because they were both SIP and IAX compatible. No documentation came with them and on the web page I could only find SLD specs, not model ALDU specs. Hm - I wonder how good that IAX claim is. http://www.act-tel.com.tw/_UpLoad/Pic/P104S-Spec.pdf Any suggestions how I should proceed? TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
The timing is provided by Zaptel, if you don't have that there is no timing. So, after we get the registration, the next step is get the TDMoE working, if not you cannot generate any calls. If I confuse you, I'm sorry but was trying to help not to make you loose. I don't use ztdummy but you should request more info in that direction to finish this. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Carlos Alperin wrote: Ok, Now you get registration, but still you cannot complete the call. I don't see your dialplan on your files. However, the error log shows that ztdummy is not working, so there is no timming. Lsmod shows the module already installed? Carlos I did have it on a previous post, but here again: (note I have no special HW so I did not modprobe zapte before modprobe ztdummy - not sure if that is relevant or not) gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # * extensions.conf * [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [geograph] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() exten = 201,1,Dial(IAX/z1,20,tr) exten = 202,1,Dial(IAX/z2,20,tr) exten = 1201,1,Dial(IAX/z1,20,tr) exten = 1202,1,Dial(IAX/z2,20,tr) *** /var/log/messages ** Jul 8 19:40:03 WARNING[6707]: Unable to open pseudo channel for timing... Sound may be choppy. Jul 8 19:40:03 WARNING[6707]: Unable to open IAX timing interface: No such file or directory Jul 8 19:40:03 NOTICE[6707]: Unable to load config sip.conf, SIP disabled Jul 8 19:40:03 WARNING[6707]: Read error on sound device: Resource temporarily unavailable Jul 8 19:40:03 WARNING[6707]: Unable to get our IP address, Skinny disabled *** ___ 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] IAXphone - ip address - extension number.
If putting a host= statement in * does anything in terms of completing the registration, there is a serious infrastructure issue. That isn't going to lead him to resolving the registration problem. Why not try 'iax2 debug' and let's take a look at what's going on to resolve the registration problem. You're right, but here the priority is why those phones are not registered. Is obvious that Asterisk doesn't get any info from the phones, and didn't found the way to reach them. Host is Unspecified, so host=ipaddress or host=name (But you should be sure your DNS works) can be easily setup to check if the registration is completed. We 'd been through this doing iax2 trunks, and we always get the same issue, until you define were your peer is, you never get the registration completed, even due to authentication problems. No offense, only trying to make it easy to get a result. Regards, Carlos -Original Message- Carlos, be careful, you've been given out either bad info or incomplete info. The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) No offense, just be careful. This is simple. You didn't define on the IAX.conf where to find the phones. You need to tell the system which IP has each phone. That was what I thought this morning when you send me the files. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Friday, July 08, 2005 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. Under Host, the IP of the phone should be shown, instead of Unspecified check the config of the phones, they are not registering with Asterisk hth ___ 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 ---End of Original Message- ___ 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 ---End of Original Message- ___ 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] IAXphone - ip address - extension number.
The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) In my reply to time bandits suggestion, that is exactly what I did and iax2 show peers now shows the IP address. *but* I still cannot dial anywhere. What you have insinuated in your comment is that Asterisk and the phone should have automatically sorted themselves out and because this is not happening, my problem is elsewhere - proven by the fact that I have hardcoded host=192.168.0.201 in the iax.conf file (whereas before I used to have host=dynamic) and it still does not work. Possibilities: 1) The iax timing error when I load asterisk (see reply to time bandit) 2) The phone itself. They are 2 new Advantage Century Telecomms Corp AC-104ALDU units which I bought because they were both SIP and IAX compatible. No documentation came with them and on the web page I could only find SLD specs, not model ALDU specs. Hm - I wonder how good that IAX claim is. http://www.act-tel.com.tw/_UpLoad/Pic/P104S-Spec.pdf Any suggestions how I should proceed? Well, if I were working with those I'd fire up Ethereal and look to see exactly what the phone was doing. If they really are iax capable, you should see at least some iax packets coming from them. Then, the real answers to why the phone doesn't work can be resolved from the packet trace. If you're not comfortable reading ethereal traces, email a copy of the trace or put it somewhere that we can drag it down. If the trace indicates nothing, then either the phone is configured incorrectly, or, its just not going to work with iax. Rich ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: If I write try with sip, everybody will try to hang me in the highest tree. :-) (you have a nice attitude) So, If was on your position, I'll try SIP. I did actually have a quick try - and got similar problems - but I'll have to have a good look to know if the probs are the same. I still think it's a ztdummy issue - is that also needed with SIP? I'm trying to help, no to argue with anyone, and of course not to dissinform or missinterpretate anything. Just help. you're cool - thanks Regards, Carlos ___ 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] IAXphone - ip address - extension number.
No one is going to hang you, Carlos. Trying sip probably isn't a bad idea at all, but that effort still wouldn't resolve the iax issue. ;) If I write try with sip, everybody will try to hang me in the highest tree. So, If was on your position, I'll try SIP. I'm trying to help, no to argue with anyone, and of course not to dissinform or missinterpretate anything. Just help. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Rich Adamson wrote: The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) In my reply to time bandits suggestion, that is exactly what I did and iax2 show peers now shows the IP address. *but* I still cannot dial anywhere. What you have insinuated in your comment is that Asterisk and the phone should have automatically sorted themselves out and because this is not happening, my problem is elsewhere - proven by the fact that I have hardcoded host=192.168.0.201 in the iax.conf file (whereas before I used to have host=dynamic) and it still does not work. Possibilities: 1) The iax timing error when I load asterisk (see reply to time bandit) 2) The phone itself. They are 2 new Advantage Century Telecomms Corp AC-104ALDU units which I bought because they were both SIP and IAX compatible. No documentation came with them and on the web page I could only find SLD specs, not model ALDU specs. Hm - I wonder how good that IAX claim is. http://www.act-tel.com.tw/_UpLoad/Pic/P104S-Spec.pdf Any suggestions how I should proceed? TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: The timing is provided by Zaptel, if you don't have that there is no timing. So, after we get the registration, the next step is get the TDMoE working, if not you cannot generate any calls. If I confuse you, I'm sorry but was trying to help not to make you loose. I don't use ztdummy but you should request more info in that direction to finish this. OK - I tried zaprtc but it wouldn't load - I'll start another thread on the merits of ztdummy vs zaprtc have a good weekend and thanks for your support. Zoltan. Regards, Carlos ___ 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] IAXphone - ip address - extension number.
If I write try with sip, everybody will try to hang me in the highest tree. :-) (you have a nice attitude) So, If was on your position, I'll try SIP. I did actually have a quick try - and got similar problems - but I'll have to have a good look to know if the probs are the same. I still think it's a ztdummy issue - is that also needed with SIP? Zaptel timing (eg, ztdummy) isn't needed for iax or sip. Its only needed on a couple of application items like the meetme, etc, which are documented on the wiki. ___ 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] IAXphone - ip address - extension number.
Ha! (again) Hopefully this helps you guys a bit (to help me ) Ta yet again, Zoltan From the wiki page about my phone, I changed my iax.conf to: (note 6 lines after CallerID) * sip.conf ** [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=192.168.0.201 mailbox=1201 context=geograph callerid=Zoltan1 201 qualify=yes disallow=all allow=ulaw allow=alaw allow=g729 dtmfmode=info [z2] type=friend host=192.168.0.202 mailbox=1202 context=geograph callerid=Zoltan2 202 qualify=yes disallow=all allow=ulaw allow=alaw allow=g729 dtmfmode=info ** end sip.conf asterisk -vvvc *** Asterisk Ready. *CLI Jul 8 19:46:21 NOTICE[6826]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 8 19:46:21 NOTICE[6826]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (S) 255.255.255.255 4569 UNREACHABLE z1 192.168.0.201 (S) 255.255.255.255 4569 UNREACHABLE *CLI Yes, the above indicates the phones did in fact register at least one time, as indicated by the IP address in the Host colume. The Status = Unreachable is saying the phones no longer are reachable by asterisk. So, either the phone is going to sleep on its own, or, for whatever reason asterisk cannot send a packet to the phone and get any reasonable response. Now is the time to do a 'iax2 debug' and reboot the phone. You should see the registration occur again, followed by asterisk trying to send the phone something (some time later). We need to see the copy/paste results of that. ___ 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] IAXphone - ip address - extension number.
I agree 100%, and as you can see now they complete the registration. The problem that was hidden is that there is no timing working on the system. So, I don't think that dialing or anything else is going to work. Now, what happen with the network, I don't know because it looks like everything is on the same subnet. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, July 08, 2005 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAXphone - ip address - extension number. If putting a host= statement in * does anything in terms of completing the registration, there is a serious infrastructure issue. That isn't going to lead him to resolving the registration problem. Why not try 'iax2 debug' and let's take a look at what's going on to resolve the registration problem. You're right, but here the priority is why those phones are not registered. Is obvious that Asterisk doesn't get any info from the phones, and didn't found the way to reach them. Host is Unspecified, so host=ipaddress or host=name (But you should be sure your DNS works) can be easily setup to check if the registration is completed. We 'd been through this doing iax2 trunks, and we always get the same issue, until you define were your peer is, you never get the registration completed, even due to authentication problems. No offense, only trying to make it easy to get a result. Regards, Carlos -Original Message- Carlos, be careful, you've been given out either bad info or incomplete info. The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) No offense, just be careful. This is simple. You didn't define on the IAX.conf where to find the phones. You need to tell the system which IP has each phone. That was what I thought this morning when you send me the files. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Friday, July 08, 2005 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. Under Host, the IP of the phone should be shown, instead of Unspecified check the config of the phones, they are not registering with Asterisk hth ___ 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 ---End of Original Message- ___ 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 ---End of Original Message- ___ 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] IAXphone - ip address - extension number.
Zoltan, Rich is right, try to use Ethereal to see what the phones are sending out. You said that you can see the phones, so watch their traffic, disregards the http (just they're web servers) and see what happening on the port that belongs to SIP IAX. You never knows what you 're going to find. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Carlos Alperin wrote: If I write try with sip, everybody will try to hang me in the highest tree. :-) (you have a nice attitude) So, If was on your position, I'll try SIP. I did actually have a quick try - and got similar problems - but I'll have to have a good look to know if the probs are the same. I still think it's a ztdummy issue - is that also needed with SIP? I'm trying to help, no to argue with anyone, and of course not to dissinform or missinterpretate anything. Just help. you're cool - thanks Regards, Carlos ___ 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] IAXphone - ip address - extension number.
Hi Rich, See debug output after your post below. Thanks, Zoltan Rich Adamson wrote: If putting a host= statement in * does anything in terms of completing the registration, there is a serious infrastructure issue. That isn't going to lead him to resolving the registration problem. Why not try 'iax2 debug' and let's take a look at what's going on to resolve the registration problem. == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI Jul 8 20:52:54 NOTICE[7790]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 8 20:52:54 NOTICE[7790]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! *CLI iax2 debug IAX2 Debugging Enabled *CLI *CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 3 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 4 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 3 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 4 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 5 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 6 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 5 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 6 DCall: 0 [192.168.0.201:4569] ___ 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] IAXphone - ip address - extension number.
Thanks, Now I know that I don't need to change my name address. (Joke) My only concern was to be sure that every device is seen each other. I only use IAX2 for trunks between servers, which pretty good, however I hear every kind of argues about why I shouldn't use. But for me, it never worked if I don't define first the right hosts. (I have my servers, half in Michigan, the others are in Texas Florida) Zoltan's problem looks more to be a timing issue, just he is not using Zaptel but ztdummy modules that are not loading. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, July 08, 2005 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAXphone - ip address - extension number. No one is going to hang you, Carlos. Trying sip probably isn't a bad idea at all, but that effort still wouldn't resolve the iax issue. ;) If I write try with sip, everybody will try to hang me in the highest tree. So, If was on your position, I'll try SIP. I'm trying to help, no to argue with anyone, and of course not to dissinform or missinterpretate anything. Just help. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Rich Adamson wrote: The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) In my reply to time bandits suggestion, that is exactly what I did and iax2 show peers now shows the IP address. *but* I still cannot dial anywhere. What you have insinuated in your comment is that Asterisk and the phone should have automatically sorted themselves out and because this is not happening, my problem is elsewhere - proven by the fact that I have hardcoded host=192.168.0.201 in the iax.conf file (whereas before I used to have host=dynamic) and it still does not work. Possibilities: 1) The iax timing error when I load asterisk (see reply to time bandit) 2) The phone itself. They are 2 new Advantage Century Telecomms Corp AC-104ALDU units which I bought because they were both SIP and IAX compatible. No documentation came with them and on the web page I could only find SLD specs, not model ALDU specs. Hm - I wonder how good that IAX claim is. http://www.act-tel.com.tw/_UpLoad/Pic/P104S-Spec.pdf Any suggestions how I should proceed? TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: I still think it's a ztdummy issue - is that also needed with SIP? Zaptel timing (eg, ztdummy) isn't needed for iax or sip. Its only needed on a couple of application items like the meetme, etc, which are documented on the wiki. You're quite right about SIP not needing zaptel timing (not sure why I asked above as I did know that) but IAX is complaining about timing (during asterisk load) and, I could be mistaken, but I'm sure I saw somewhere that IAX needs the timing stuff. == Parsing '/etc/asterisk/agents.conf': Found [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Jul 8 21:09:11 WARNING[8096]: chan_iax2.c:7477 load_module: *Unable to open IAX timing interface:* No such file or directory == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [chan_local.so] = (Local Proxy Channel) But, let's kill this damn registration issue - I'll do what you asked (rebooting thephone) and send the cutpaste in a few minutes. Cheers, Zoltan ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: Well, if I were working with those I'd fire up Ethereal and look to see exactly what the phone was doing. If they really are iax capable, you should see at least some iax packets coming from them. Then, the real answers to why the phone doesn't work can be resolved from the packet trace. If you're not comfortable reading ethereal traces, email a copy of the trace or put it somewhere that we can drag it down. If the trace indicates nothing, then either the phone is configured incorrectly, or, its just not going to work with iax. Rich Hi, I've never used ethereal so have no clue about it, so... I fired up etheral gui - left everything to default. clicked start capture fired up asterisk waited for initial notice that peers are UNREACHABLE started iax2 debug waited for a batch of messages stopped ethereal. Herewith the saved ethereal file. (but I noticed nothing from 192.168.0.201 or 202 in the gui window :-( ) TIA, Zoltan PS: whats the bet this list won't allow an attachment file... (if so, do you have anywhere I can ftp it to?) ethereal_20050708_21h10 Description: Binary data ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: Yes, the above indicates the phones did in fact register at least one time, as indicated by the IP address in the Host colume. The Status = Unreachable is saying the phones no longer are reachable by asterisk. So, either the phone is going to sleep on its own, or, for whatever reason asterisk cannot send a packet to the phone and get any reasonable response. Now is the time to do a 'iax2 debug' and reboot the phone. You should see the registration occur again, followed by asterisk trying to send the phone something (some time later). We need to see the copy/paste results of that. * I pulled the power out of z2 before starting asterisk (z1 untouched and on)** == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI Jul 8 22:08:18 NOTICE[8913]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 8 22:08:18 NOTICE[8913]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! *CLI iax2 debug IAX2 Debugging Enabled *CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 3 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 1ms SCall: 4 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 3 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 1ms SCall: 4 DCall: 0 [192.168.0.201:4569] *** I put the power back on to z2 * *CLI *CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 5 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 6 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 5 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 6 DCall: 0 [192.168.0.201:4569] ** z2 finished rebooting * *CLI *CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 6ms SCall: 7 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 6ms SCall: 8 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 6ms SCall: 7 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 6ms SCall: 8 DCall: 0 [192.168.0.201:4569] *CLI *CLI iax2 debugTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 9ms SCall: 9 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 9ms SCall: 00010 DCall: 0 [192.168.0.201:4569] stop nowTx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 9ms SCall: 9 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 9ms SCall: 00010 DCall: 0 [192.168.0.201:4569] Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). gl0:/home/zls # gl0:/home/zls # ** did a few pings for peace of mind * gl0:/home/zls # ping 192.168.0.201 PING 192.168.0.201 (192.168.0.201) 56(84) bytes of data. 64 bytes from 192.168.0.201: icmp_seq=1 ttl=60 time=1.34 ms 64 bytes from 192.168.0.201: icmp_seq=2 ttl=60 time=1.58 ms --- 192.168.0.201 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 1.344/1.462/1.580/0.118 ms gl0:/home/zls # ping 192.168.0.202 PING 192.168.0.202 (192.168.0.202) 56(84) bytes of data. 64 bytes from 192.168.0.202: icmp_seq=1 ttl=60 time=1.26 ms 64 bytes from 192.168.0.202: icmp_seq=2 ttl=60 time=1.42 ms --- 192.168.0.202 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1000ms rtt min/avg/max/mdev = 1.260/1.342/1.425/0.090 ms gl0:/home/zls # ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: Zoltan's problem looks more to be a timing issue, just he is not using Zaptel but ztdummy modules that are not loading. Regards, Carlos Is this how the modprobes are supposed to respond?? gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # ___ 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] IAXphone - ip address - extension number.
Zoltan, The debug stuff at the bottom of this email looks normal. Do the 'iax2 debug' again, but include the registration (reboot the phone), copy/paste that, then let the debug run until something different then those POKES show up. To save a little time, dial from one iax phone to another and copy/paste that into an email as well. All of the above should give a very good clue on the issues to be resolved. Rich See debug output after your post below. Thanks, Zoltan Rich Adamson wrote: If putting a host= statement in * does anything in terms of completing the registration, there is a serious infrastructure issue. That isn't going to lead him to resolving the registration problem. Why not try 'iax2 debug' and let's take a look at what's going on to resolve the registration problem. == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI Jul 8 20:52:54 NOTICE[7790]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 8 20:52:54 NOTICE[7790]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! *CLI iax2 debug IAX2 Debugging Enabled *CLI *CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 3 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 4 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 3 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 4ms SCall: 4 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 5 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 6 DCall: 0 [192.168.0.201:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 5 DCall: 0 [192.168.0.202:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 6 DCall: 0 [192.168.0.201:4569] ___ 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] IAXphone - ip address - extension number.
Rich Adamson wrote: Yes, the above indicates the phones did in fact register at least one time, as indicated by the IP address in the Host colume. The Status = Unreachable is saying the phones no longer are reachable by asterisk. So, either the phone is going to sleep on its own, or, for whatever reason asterisk cannot send a packet to the phone and get any reasonable response. Now is the time to do a 'iax2 debug' and reboot the phone. You should see the registration occur again, followed by asterisk trying to send the phone something (some time later). We need to see the copy/paste results of that. I was browsing through the phone menus (browser to 192.168.0.201:) and I found a few settings that I fiddled with: 1) Protocol: SIP, IAX or BOTH (both was set so I left it so) 2) Phone Display Number (was SIP, I changed it to IAX) 3) IAX dial prefix: (was set to *1 but I wasn't using it) 4) IAX Server (was blank, I set it to asterisk: 192.168.0.100) 5)IAX Server port local port are (and were) both at 4569 6) IAX Number - still left blank 7) IAXname - still left blank 8) IAX username - still left blank 9) IAX password - still left blank 10) IAX refresh interval - still left as 60 seconds 11) UPnP - still left disabled *but* even though I still cannot get the other phone to ring, look at asterisks response: (I now dial *1202 instead of just 202 - if just 202 then asterisk does not respond at all) Asterisk Ready. *CLI Jul 8 23:23:11 NOTICE[10202]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z2' is now UNREACHABLE! Jul 8 23:23:11 NOTICE[10202]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'z1' is now UNREACHABLE! Jul 8 23:23:13 NOTICE[10202]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 8 23:23:18 NOTICE[10202]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 -- Accepting unauthenticated call from 192.168.0.202, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/7, IAX/z1|20|tr) in new stack Jul 8 23:23:24 WARNING[10202]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 8 23:23:24 NOTICE[10202]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 8 23:23:34 WARNING[10202]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/7' Jul 8 23:23:45 NOTICE[10202]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 8 23:23:50 NOTICE[10202]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 Jul 8 23:23:50 NOTICE[10202]: chan_iax2.c:5468 socket_read: Rejected connect attempt from 192.168.0.201, request '[EMAIL PROTECTED]' does not exist -- Accepting unauthenticated call from 192.168.0.201, requested format = 4, actual format = 4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/17, IAX/z2|20|tr) in new stack Jul 8 23:24:06 WARNING[10202]: channel.c:1913 ast_request: No channel type registered for 'IAX' Jul 8 23:24:06 NOTICE[10202]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy/congested at this time Jul 8 23:24:16 WARNING[10202]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'geograph' -- Hungup 'IAX2/[EMAIL PROTECTED]/17' Jul 8 23:24:17 NOTICE[10202]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.201 Jul 8 23:24:22 NOTICE[10202]: chan_iax2.c:3891 register_verify: Empty registration from 192.168.0.202 *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). gl0:/home/zls # ___ 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] IAXphone - ip address - extension number.
Well, if I were working with those I'd fire up Ethereal and look to see exactly what the phone was doing. If they really are iax capable, you should see at least some iax packets coming from them. Then, the real answers to why the phone doesn't work can be resolved from the packet trace. If you're not comfortable reading ethereal traces, email a copy of the trace or put it somewhere that we can drag it down. If the trace indicates nothing, then either the phone is configured incorrectly, or, its just not going to work with iax. Rich Hi, I've never used ethereal so have no clue about it, so... I fired up etheral gui - left everything to default. clicked start capture fired up asterisk waited for initial notice that peers are UNREACHABLE started iax2 debug waited for a batch of messages stopped ethereal. Herewith the saved ethereal file. (but I noticed nothing from 192.168.0.201 or 202 in the gui window :-( ) TIA, Zoltan PS: whats the bet this list won't allow an attachment file... (if so, do you have anywhere I can ftp it to?) The attachment (trace) came through just fine. It was smaller then some people's email signatures. ;) I looked through the trace and do not see any communications to those phones at all. That's likely because you ran ethereal on a different system then asterisk is running on, and the NetGear switch that you are using only passes broadcast packets (expected). Try it again, but this time run ethereal on your asterisk box. If that box has two nic interfaces, when starting the capture choose the nic interface that has the phones on it. It might take an extra attempt or two, but you should be able to see the IP addresses of your phones in that ethereal display. If the trace becomes rather large, just email it directly to me so we don't impact the asterisk list. Rich ___ 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] IAXphone - ip address - extension number.
I don't see any packet coming from 192.168.0.201 192.168.0.202. But I see also packets from other networks plus spanning tree protocol, like if you have a switch with STP... Is your computer on the same network that the phones? Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 3:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Rich Adamson wrote: Well, if I were working with those I'd fire up Ethereal and look to see exactly what the phone was doing. If they really are iax capable, you should see at least some iax packets coming from them. Then, the real answers to why the phone doesn't work can be resolved from the packet trace. If you're not comfortable reading ethereal traces, email a copy of the trace or put it somewhere that we can drag it down. If the trace indicates nothing, then either the phone is configured incorrectly, or, its just not going to work with iax. Rich Hi, I've never used ethereal so have no clue about it, so... I fired up etheral gui - left everything to default. clicked start capture fired up asterisk waited for initial notice that peers are UNREACHABLE started iax2 debug waited for a batch of messages stopped ethereal. Herewith the saved ethereal file. (but I noticed nothing from 192.168.0.201 or 202 in the gui window :-( ) TIA, Zoltan PS: whats the bet this list won't allow an attachment file... (if so, do you have anywhere I can ftp it to?) ___ 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] IAXphone - ip address - extension number.
Yes, But it looks like they cannot open a device on the Zaptel.conf. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Carlos Alperin wrote: Zoltan's problem looks more to be a timing issue, just he is not using Zaptel but ztdummy modules that are not loading. Regards, Carlos Is this how the modprobes are supposed to respond?? gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # ___ 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] IAXphone - ip address - extension number.
On Fri, Jul 08, 2005 at 11:12:37PM +0200, Zoltan Szecsei wrote: Is this how the modprobes are supposed to respond?? gl0:/home/zls # modprobe zaptel gl0:/home/zls # lsmod | grep z Module Size Used by zaptel239620 0 crc_ccitt 6144 1 zaptel gl0:/home/zls # modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 142: Unable to open master device '/dev/zap/ctl' gl0:/home/zls # lsmod | grep z Module Size Used by ztdummy 7744 0 zaptel239620 1 ztdummy crc_ccitt 6144 1 zaptel gl0:/home/zls # This error does not come from the kernel when initilizing the module. Rather, it comes from the post-install user-space action of 'ztcfg'. ztdummy does not need the insmod postinstall action of ztcfg, beccause there is nothing to configure in it (/me too lazy to file a bug report) However if you don't have the ztcfg device files they need to be created. Do you use kernel 2.4 or 2.6? If you use 2.4, consider 2.6, as its ztdummy works better. If you use 2.6, you may be using udev, and need to read README.udev . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: The only reason for try SIP, is to find where the problem is. You can use what you prefer, if you can made it works. Any chance to see both SIP IAX.conf? Thanks, Carlos Hi Carlos, Thanks for you help thusfar. I have provided: iax.conf extensions.conf bits of log from asterisk -vvgc last bits from making zaptel modprobe/lsmod ztdummy. Note that I have renamed zaptel.conf and sip.conf because I am under the impression I do not need them. I dont want to add SIP until IAX works, and I have no special HW, so dont need zaptel.conf. Note also at this stage extensions.conf has only what I hope it needs to just make the other phone ring. The SuSE 9.3 box running asterisk 1.0.9 has IP 192.168.0.100 and phones z1 and z2 have 192.168.0.201 and 202 respectively. ++ gl0:/etc/asterisk # cat iax.conf [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=dynamic mailbox=1201 context=geograph callerid=Zoltan1 201 [z2] type=friend host=dynamic mailbox=1202 context=geograph callerid=Zoltan2 202 gl0:/etc/asterisk # ++ gl0:/etc/asterisk # cat extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:[EMAIL PROTECTED] [geograph] exten= 201,1,Dial(IAX/z1,20,tr) exten= 202,1,Dial(IAX/z2,20,tr) exten= 1201,1,Dial(IAX/z1,20,tr) ; not relevant yet exten= 1202,1,Dial(IAX/z2,20,tr) ; not relevant yet ++ From asterisk -vvvgc snip===8===snip===8===snip===8===snip=== [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Jul 8 11:58:13 WARNING[32206]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [pbx_config.so] = (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found -- Setting global variable 'CONSOLE' to 'Console/dsp' -- Setting global variable 'IAXINFO' to 'guest' -- Setting global variable 'TRUNK' to 'Zap/g2' -- Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'geograph' -- Added extension '201' priority 1 to geograph -- Added extension '202' priority 1 to geograph -- Added extension '1201' priority 1 to geograph -- Added extension '1202' priority 1 to geograph [pbx_spool.so] = (Outgoing Spool Support) [cdr_csv.so] = (Comma Separated Values CDR Backend) [cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found [chan_agent.so] = (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' == Registered application 'AgentMonitorOutgoing' == Parsing '/etc/asterisk/agents.conf': Found [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Jul 8 11:58:13 WARNING[32206]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such file or directory == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) snip===8===snip===8===snip===8===snip=== [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Jul 8 11:58:13 NOTICE[32206]: chan_sip.c:8679 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_skinny.so]Jul 8 11:58:13 WARNING[32206]: chan_oss.c:257 sound_thread: Read error on sound device: Resource temporarily unavailable = (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found Jul 8 11:58:13 WARNING[32206]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [codec_a_mu.so] = (A-law and Mulaw direct Coder/Decoder) ==
RE: [Asterisk-Users] IAXphone - ip address - extension number.
Zoltan, Ok. I believe that I know what is going on: Just for confirm can you go inside the system : run asterisk -r Then run IAX2 SHOW PEERS and tell me what you get? When you finish you can exit with quit. I think that the system doesn't know where are the phones located. Thanks, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 6:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Carlos Alperin wrote: The only reason for try SIP, is to find where the problem is. You can use what you prefer, if you can made it works. Any chance to see both SIP IAX.conf? Thanks, Carlos Hi Carlos, Thanks for you help thusfar. I have provided: iax.conf extensions.conf bits of log from asterisk -vvgc last bits from making zaptel modprobe/lsmod ztdummy. Note that I have renamed zaptel.conf and sip.conf because I am under the impression I do not need them. I dont want to add SIP until IAX works, and I have no special HW, so dont need zaptel.conf. Note also at this stage extensions.conf has only what I hope it needs to just make the other phone ring. The SuSE 9.3 box running asterisk 1.0.9 has IP 192.168.0.100 and phones z1 and z2 have 192.168.0.201 and 202 respectively. ++ gl0:/etc/asterisk # cat iax.conf [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=dynamic mailbox=1201 context=geograph callerid=Zoltan1 201 [z2] type=friend host=dynamic mailbox=1202 context=geograph callerid=Zoltan2 202 gl0:/etc/asterisk # ++ gl0:/etc/asterisk # cat extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:[EMAIL PROTECTED] [geograph] exten= 201,1,Dial(IAX/z1,20,tr) exten= 202,1,Dial(IAX/z2,20,tr) exten= 1201,1,Dial(IAX/z1,20,tr) ; not relevant yet exten= 1202,1,Dial(IAX/z2,20,tr) ; not relevant yet ++ From asterisk -vvvgc snip===8===snip===8===snip===8===snip=== [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Jul 8 11:58:13 WARNING[32206]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [pbx_config.so] = (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found -- Setting global variable 'CONSOLE' to 'Console/dsp' -- Setting global variable 'IAXINFO' to 'guest' -- Setting global variable 'TRUNK' to 'Zap/g2' -- Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'geograph' -- Added extension '201' priority 1 to geograph -- Added extension '202' priority 1 to geograph -- Added extension '1201' priority 1 to geograph -- Added extension '1202' priority 1 to geograph [pbx_spool.so] = (Outgoing Spool Support) [cdr_csv.so] = (Comma Separated Values CDR Backend) [cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found [chan_agent.so] = (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' == Registered application 'AgentMonitorOutgoing' == Parsing '/etc/asterisk/agents.conf': Found [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Jul 8 11:58:13 WARNING[32206]: chan_iax2.c:7477 load_module: Unable to open IAX timing interface: No such file or directory == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) snip===8===snip===8===snip===8===snip=== [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Jul 8 11:58:13 NOTICE[32206]: chan_sip.c:8679 reload_config: Unable to load config sip.conf, SIP disabled
Re: [Asterisk-Users] IAXphone - ip address - extension number.
Hi, Sorry - just got back. Here is what you wanted, but I am also concerned about the unable to open iax timing interface - scan this email for timing to pick it up below. Note the lsmod at the very end of this email, showing that ztdummy is loaded. I tried compiling and loading zaprtc after I sent the previous email, but I could not successfully make load it. Thanks for helping, Zoltan. + == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored *CLI *CLI *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). gl0:/home/zls # +++ Carlos Alperin wrote: Zoltan, Ok. I believe that I know what is going on: Just for confirm can you go inside the system : run asterisk -r Then run IAX2 SHOW PEERS and tell me what you get? When you finish you can exit with quit. I think that the system doesn't know where are the phones located. Thanks, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 6:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Carlos Alperin wrote: The only reason for try SIP, is to find where the problem is. You can use what you prefer, if you can made it works. Any chance to see both SIP IAX.conf? Thanks, Carlos Hi Carlos, Thanks for you help thusfar. I have provided: iax.conf extensions.conf bits of log from asterisk -vvgc last bits from making zaptel modprobe/lsmod ztdummy. Note that I have renamed zaptel.conf and sip.conf because I am under the impression I do not need them. I dont want to add SIP until IAX works, and I have no special HW, so dont need zaptel.conf. Note also at this stage extensions.conf has only what I hope it needs to just make the other phone ring. The SuSE 9.3 box running asterisk 1.0.9 has IP 192.168.0.100 and phones z1 and z2 have 192.168.0.201 and 202 respectively. ++ gl0:/etc/asterisk # cat iax.conf [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=dynamic mailbox=1201 context=geograph callerid=Zoltan1 201 [z2] type=friend host=dynamic mailbox=1202 context=geograph callerid=Zoltan2 202 gl0:/etc/asterisk # ++ gl0:/etc/asterisk # cat extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:[EMAIL PROTECTED] [geograph] exten= 201,1,Dial(IAX/z1,20,tr) exten= 202,1,Dial(IAX/z2,20,tr) exten= 1201,1,Dial(IAX/z1,20,tr) ; not relevant yet exten= 1202,1,Dial(IAX/z2,20,tr) ; not relevant yet ++ From asterisk -vvvgc snip===8===snip===8===snip===8===snip=== [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Jul 8 11:58:13 WARNING[32206]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [pbx_config.so] = (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found -- Setting global variable 'CONSOLE' to 'Console/dsp' -- Setting global variable 'IAXINFO' to 'guest' -- Setting global variable 'TRUNK' to 'Zap/g2' -- Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'geograph' -- Added extension '201' priority 1 to geograph -- Added extension '202' priority 1 to geograph -- Added extension '1201' priority 1 to geograph -- Added extension '1202' priority 1 to geograph [pbx_spool.so] = (Outgoing Spool Support) [cdr_csv.so] = (Comma Separated Values CDR Backend) [cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found [chan_agent.so] = (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' == Registered application
Re: [Asterisk-Users] IAXphone - ip address - extension number.
*CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. Under Host, the IP of the phone should be shown, instead of Unspecified check the config of the phones, they are not registering with Asterisk hth ___ 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] IAXphone - ip address - extension number.
This is simple. You didn't define on the IAX.conf where to find the phones. You need to tell the system which IP has each phone. That was what I thought this morning when you send me the files. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Friday, July 08, 2005 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. Under Host, the IP of the phone should be shown, instead of Unspecified check the config of the phones, they are not registering with Asterisk hth ___ 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] IAXphone - ip address - extension number.
Put the IP addresses of each phone on the IAX.conf section. Then we can check the rest. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Hi, Sorry - just got back. Here is what you wanted, but I am also concerned about the unable to open iax timing interface - scan this email for timing to pick it up below. Note the lsmod at the very end of this email, showing that ztdummy is loaded. I tried compiling and loading zaprtc after I sent the previous email, but I could not successfully make load it. Thanks for helping, Zoltan. + == RTP Allocating from port range 1 - 2 Asterisk Ready. *CLI *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored *CLI *CLI *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). gl0:/home/zls # +++ Carlos Alperin wrote: Zoltan, Ok. I believe that I know what is going on: Just for confirm can you go inside the system : run asterisk -r Then run IAX2 SHOW PEERS and tell me what you get? When you finish you can exit with quit. I think that the system doesn't know where are the phones located. Thanks, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Friday, July 08, 2005 6:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. Carlos Alperin wrote: The only reason for try SIP, is to find where the problem is. You can use what you prefer, if you can made it works. Any chance to see both SIP IAX.conf? Thanks, Carlos Hi Carlos, Thanks for you help thusfar. I have provided: iax.conf extensions.conf bits of log from asterisk -vvgc last bits from making zaptel modprobe/lsmod ztdummy. Note that I have renamed zaptel.conf and sip.conf because I am under the impression I do not need them. I dont want to add SIP until IAX works, and I have no special HW, so dont need zaptel.conf. Note also at this stage extensions.conf has only what I hope it needs to just make the other phone ring. The SuSE 9.3 box running asterisk 1.0.9 has IP 192.168.0.100 and phones z1 and z2 have 192.168.0.201 and 202 respectively. ++ gl0:/etc/asterisk # cat iax.conf [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=dynamic mailbox=1201 context=geograph callerid=Zoltan1 201 [z2] type=friend host=dynamic mailbox=1202 context=geograph callerid=Zoltan2 202 gl0:/etc/asterisk # ++ gl0:/etc/asterisk # cat extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:[EMAIL PROTECTED] [geograph] exten= 201,1,Dial(IAX/z1,20,tr) exten= 202,1,Dial(IAX/z2,20,tr) exten= 1201,1,Dial(IAX/z1,20,tr) ; not relevant yet exten= 1202,1,Dial(IAX/z2,20,tr) ; not relevant yet ++ From asterisk -vvvgc snip===8===snip===8===snip===8===snip=== [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Jul 8 11:58:13 WARNING[32206]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [pbx_config.so] = (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found -- Setting global variable 'CONSOLE' to 'Console/dsp' -- Setting global variable 'IAXINFO' to 'guest' -- Setting global variable 'TRUNK' to 'Zap/g2' -- Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'geograph' -- Added extension '201' priority 1 to geograph -- Added extension '202' priority 1 to geograph -- Added extension '1201' priority 1 to geograph -- Added extension '1202' priority 1 to geograph [pbx_spool.so] = (Outgoing Spool Support
RE: [Asterisk-Users] IAXphone - ip address - extension number.
Carlos, be careful, you've been given out either bad info or incomplete info. The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) No offense, just be careful. This is simple. You didn't define on the IAX.conf where to find the phones. You need to tell the system which IP has each phone. That was what I thought this morning when you send me the files. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Friday, July 08, 2005 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. Under Host, the IP of the phone should be shown, instead of Unspecified check the config of the phones, they are not registering with Asterisk hth ___ 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 ---End of Original Message- ___ 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] IAXphone - ip address - extension number.
Time Bandit wrote: *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. OK, my iax.conf now has: [general] port=4569 bindaddr=0.0.0.0 bandwidth=medium disallow=LPC10 ;deny=all ;all0w=ulaw ;all0w=alaw ;all0w=gsm jitterbuffer=yes ; register = zoltan:[EMAIL PROTECTED] [z1] type=friend host=192.168.0.201 mailbox=1201 context=geograph callerid=Zoltan1 201 [z2] type=friend host=192.168.0.202 mailbox=1202 context=geograph callerid=Zoltan2 202 **Then I load asterisk -vvvc: Asterisk Ready. *CLI *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 192.168.0.202 (S) 255.255.255.255 4569 Unmonitored z1 192.168.0.201 (S) 255.255.255.255 4569 Unmonitored *CLI *CLI *CLI iax2 show users Username SecretAuthen Def.Context A/C z2 -no secret- 003 geograph No z1 -no secret- 003 geograph No *CLI *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). gl0:/home/zls # ** Dialing from one phone ** on phone z1 I go offhook (hear dialtone); dial 202 and get enagaged tone. Similar if I dial 1, or 2, or 201, or 202 or anything in fact. If I go through the menu on the handset, these are the correct IP addresses of the phones. I can ping these phone IP addresses from any other PC on the network. *My extensions.conf is:*** [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [geograph] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() exten = 201,1,Dial(IAX/z1,20,tr) exten = 202,1,Dial(IAX/z2,20,tr) exten = 1201,1,Dial(IAX/z1,20,tr) exten = 1202,1,Dial(IAX/z2,20,tr) The last 4 entries in /var/log/asterisk/messages is: * Jul 8 17:57:32 WARNING[5219]: Unable to open pseudo channel for timing... Sound may be choppy. Jul 8 17:57:32 WARNING[5219]: Unable to open IAX timing interface: No such file or directory Jul 8 17:57:32 NOTICE[5219]: Unable to load config sip.conf, SIP disabled Jul 8 17:57:32 WARNING[5219]: Unable to get our IP address, Skinny disabled Jul 8 17:57:32 WARNING[5219]: Read error on sound device: Resource temporarily unavailable My head is: Bald please help. TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
You're right, but here the priority is why those phones are not registered. Is obvious that Asterisk doesn't get any info from the phones, and didn't found the way to reach them. Host is Unspecified, so host=ipaddress or host=name (But you should be sure your DNS works) can be easily setup to check if the registration is completed. We 'd been through this doing iax2 trunks, and we always get the same issue, until you define were your peer is, you never get the registration completed, even due to authentication problems. No offense, only trying to make it easy to get a result. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, July 08, 2005 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAXphone - ip address - extension number. Carlos, be careful, you've been given out either bad info or incomplete info. The iax2 show peers is indicating the phone is not registering properly. (If the phone never changes IP addresses ever, then he could put a host= statement in the iax.conf, but that is very non-standard and will only serve to confuse people.) No offense, just be careful. This is simple. You didn't define on the IAX.conf where to find the phones. You need to tell the system which IP has each phone. That was what I thought this morning when you send me the files. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Friday, July 08, 2005 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. *CLI iax2 show peers Name/UsernameHost Mask Port Status z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored From this, you can see that none of the IAX phones are registered. Under Host, the IP of the phone should be shown, instead of Unspecified check the config of the phones, they are not registering with Asterisk hth ___ 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 ---End of Original Message- ___ 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] IAXphone - ip address - extension number.
Hi, I'm trying to set up two ACT SIP/IAX capable phones to communicate with each other on the same internal network, using asterisk 1.0.9 on SuSE 9.3 (because I intend to grow the situation after this basic setup is functioning) The phone IPs are set to 192.168.0.201 and 202 respectively. I've had a look at iax.conf and extensions.conf but cannot see how to tie these IPs to an extension number, let alone how to dial that extension. The Getting Started with Asterisk and the Asterisk Doc Proj - Vol 1 that I have been using just have far too much info to work out what can be ignored in order to get such a simple setup working. I'd be happy for any help or pointers to steps that I should have followed. TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
What about define those phones on the SIP.conf and use sip, instead of IAX. That protocol use be more used to communicate Asterisk servers more than phones. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Thursday, July 07, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAXphone - ip address - extension number. Hi, I'm trying to set up two ACT SIP/IAX capable phones to communicate with each other on the same internal network, using asterisk 1.0.9 on SuSE 9.3 (because I intend to grow the situation after this basic setup is functioning) The phone IPs are set to 192.168.0.201 and 202 respectively. I've had a look at iax.conf and extensions.conf but cannot see how to tie these IPs to an extension number, let alone how to dial that extension. The Getting Started with Asterisk and the Asterisk Doc Proj - Vol 1 that I have been using just have far too much info to work out what can be ignored in order to get such a simple setup working. I'd be happy for any help or pointers to steps that I should have followed. TIA, Zoltan ___ 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] IAXphone - ip address - extension number.
Carlos Alperin wrote: What about define those phones on the SIP.conf and use sip, instead of IAX. That protocol use be more used to communicate Asterisk servers more than phones. Regards, Carlos Alperin Ah - ok - I understood from the docs that IAX was better and, as the phone was capable of both, I've been trying to get it going via IAX. regards, Zoltan ___ 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] IAXphone - ip address - extension number.
What about define those phones on the SIP.conf and use sip, instead of IAX. That protocol use be more used to communicate Asterisk servers more than phones. That's not totally true. An IAX softphone will work easily behind a NAT/Firewall. The same can't be said for a SIP one. I've tested IAX succesfully working behind 3 NAT, and all the 4 softphones where able to place/receive calls. I really don't think you could do the same with SIP. Ah - ok - I understood from the docs that IAX was better and, as the phone was capable of both, I've been trying to get it going via IAX. My opinion is to keep using IAX, because, like you concluded, it's a better protocol. hth ___ 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] IAXphone - ip address - extension number.
My opinion is to keep using IAX, because, like you concluded, it's a better protocol. hth hth? - well, only if you can give me some pointers as to what I should be looking at to make it work :-) (all right, yes it does help: you've given me confidence in my conviction, but not helped in making me realise where I've been dorf. :-) ) zoltan. ___ 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] IAXphone - ip address - extension number.
The only reason for try SIP, is to find where the problem is. You can use what you prefer, if you can made it works. Any chance to see both SIP IAX.conf? Thanks, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Thursday, July 07, 2005 6:01 PM To: Time Bandit; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXphone - ip address - extension number. My opinion is to keep using IAX, because, like you concluded, it's a better protocol. hth hth? - well, only if you can give me some pointers as to what I should be looking at to make it work :-) (all right, yes it does help: you've given me confidence in my conviction, but not helped in making me realise where I've been dorf. :-) ) zoltan. ___ 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] Iaxphone - unreachable if qualify yes ?
Hi, if I change Iaxphone settings to qualify=yes it says it's unreachable. Can iax2 clients be monitored with qualify option ? Is this problem related to iaxphone ? Anyone sucessfully using iax qualify feature ? Regards, Rob. ___ 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] Iaxphone - unreachable if qualify yes ?
Robert Rozman wrote: Hi, if I change Iaxphone settings to qualify=yes it says it's unreachable. Can iax2 clients be monitored with qualify option ? Is this problem related to iaxphone ? Anyone sucessfully using iax qualify feature ? It was just fixed in iaxclient-cvs this week, I think.. Don't know if/when Steve Sokol plans to build new iaxphone binaries.. -SteveK ___ 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] IAXPHONE failures in calls to Cisco Phones
I have been operating a functional asterisk system using Fedora in a 500 MHz Pentium III Stations are Cisco 7960s and Grandstream 102s. We needed to identify a software based phone to handle traveling users, so we tried IAXPHONE's latest version. Interestingly, calls from the IAX client to 7960 phones fail. going right to voicemail. The following lists the console messages from one such attempt. [Note the Response 400 Bad request.]: -- Accepting AUTHENTICATED call from 68.106.146.79, requested format = 2, actual format = 2 -- Executing Macro([EMAIL PROTECTED]/9, exten-vm|6410|6410) in new stack -- Executing SetMusicOnHold([EMAIL PROTECTED]/9, default) in new stack -- Executing Dial([EMAIL PROTECTED]/9, SIP/6410|20|tr) in new stack -- Called 6410 -- Got SIP response 400 Bad Request back from 68.106.146.79 == No one is available to answer at this time -- Executing VoiceMail2([EMAIL PROTECTED]/9, u6410) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing 'digits/4' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') If, however, we use the IAX client to call a Grandstream phone, the call works. Here are the coinsole message from such a call: -- Accepting AUTHENTICATED call from 68.106.146.79, requested format = 2, actual format = 2 -- Executing Macro([EMAIL PROTECTED]/3, exten-vm|6413|6413) in new stack -- Executing SetMusicOnHold([EMAIL PROTECTED]/3, default) in new stack -- Executing Dial([EMAIL PROTECTED]/3, SIP/6413|20|tr) in new stack -- Called 6413 -- SIP/6413-9405 is ringing -- SIP/6413-9405 answered [EMAIL PROTECTED]/3 == Spawn extension (macro-exten-vm, s, 2) exited non-zero on '[EMAIL PROTECTED]/3' in macro 'exten-vm' == Spawn extension (from-iaxphone, 6413, 1) exited non-zero on '[EMAIL PROTECTED]/3' -- Hungup '[EMAIL PROTECTED]/3' The strange thing is, that both extensions are defined identically in extensions.conf file: ; User Name exten = 6410,1,Macro(exten-vm,6410,6410) exten = 6411,1,Macro(exten-vm,6411,6411) exten = 6412,1,Macro(exten-vm,6412,6412) exten = 6413,1,Macro(exten-vm,6413,6413) exten = ,1,Macro(exten-vm,,) Has anyone else experienced this, or does anyone have idea what's wrong? Many thanks! ___ 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] IAXPHONE failures in calls to Cisco Phones
MLS Drop for SysAdmin wrote: Has anyone else experienced this, or does anyone have idea what's wrong? Check your codecs. I'm not intimately familiar with the 7960s, but it sure smells like that is a likely possibility. B. ___ 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] Iaxphone problem
Hi, Anyone, thathas been workingwith IaxPhone http://www.sokol-associates.com/IaxPhone.htm? I have a rejected problem (maybe something with the password). But, DIAX with the same conf files works fine. Any help? Thanks, Marin Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online
