Re: [asterisk-users] PRI commands missing...
Well, this happened to me one time when I forgot to compile the pri library before the asterisk! Could you have done that? on Wednesday 10/31/2007 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Wed, Oct 31, 2007 at 12:06:25AM -0600, Carlos Chavez wrote: I have an Asterisk server running Elastix but patched to use Unicall. Everything seems to be working fine and the TE220 card is up and running with port 1 configured as PRI and port 2 as MFC/R2. We can already send and receive calls on port two but we cannot on port one. That is when we noticed that there are no PRI commands available on the Asterisk CLI. We cannot use PRI DEBUG SPAN to determine why port 1 is not receiving or sending calls. Why would this commands be missing? I wonder how those two should interact. The first thing chan_zap tries to do is to open all of its spans. Maybe it has failed there? Try playing with [trunkgroups] to explicitly tell it to only touch the Zaptel spans that are PRI. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxing with Asterisk SpanDSP [Was Fax Problems with SpanDSP]
Steve Underwood wrote: snip / ... SpanDSP handles faxes within Asterisk, through the app_rxfax and app_txfax applications. It handles faxes outside Asterisk when used with iaxmodem (there is actually a copy inside the iaxmodem package). SpanDSP cannot be used by the standard distribution of Asterisk, as it is GPL code. However, if you are using Asterisk within the restrictions of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily. Regards, Steve Thanks Steve, I was a bit lazy when I posted that question sorry - I just got a bit excited. A quick google and looking around the voip-info.org site pointed me to some interesting pages. Could anyone who has experience confirm if: * this will work with an x100p card so I can have a single FXO line that can detect incoming faxes and/or voice calls and enable me to use it for outgoing voice and fax? Also, I read somewhere that Asterisk should never be used with fax. IIRC correctly it was by the moderator on the Trixbox forum discussion about the trial of OSLEC. Yes, here: http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems I really don't know how many times I need to say this but you should never run faxes through Asterisk. It was not designed to handle it. -- Kerry Garrison trixbox Community Director Is he right? The question that prompted this reply was about how to turn off echo cancellation for fax traffic. Is this achievable? Many thanks Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] flooded by Maximum trunk data space exceeded messages
Hi, Using 1.4.13 and trunking a single iax channel to a similar box my asterisk console is flooded with: [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space exceeded to xx.xx.xx.xx:4569 Known issue? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
Douglas Garstang wrote: I guess... it shouldn't be too hard to find the time out value in the source and change it If you find the line, please let me know where. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] segfault - asterisk crash and restart
Rilawich Ango wrote:
Hi all,
Recently, I have upgraded the asterisk as following.
asterisk-1.4.13
asterisk-addon-1.4.4
libpri-1.4.1
zaptel-1.4.5.1
Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh
After upgrade, the server get segfault randomly and asterisk crash
and restart itself. I got 2 core dumps of the segfault. Based on the
core dump, we can't figure out the root cause to the problem as the
content of the core dump is not the same. We have no idea what the
problem is. Anyone can give me some advices.
--core dump 1--
(gdb) bt full
#0 0x0037e806e1f3 in _int_free () from /lib64/libc.so.6
No symbol table info available.
#1 0x0037e8071fac in free () from /lib64/libc.so.6
No symbol table info available.
#2 0x0046b7b7 in ast_frame_free (fr=0x1b9da4b0, cache=0)
at frame.c:369
No locals.
#3 0x2aaab1173573 in mixmonitor_thread (obj=0x1bb08220)
from /usr/lib/asterisk/modules/app_mixmonitor.so
This is clearly mixmonitor-related. I suggest you to look for similar
mixmonitor bugs in digium's mantis - if there's none, create and attach
this backtrace.
[snip]
--core dump 2--
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0 0x0044da80 in ast_var_name (var=0x10f1d58a0) at chanvars.c:69
69if (name[0] == '_') {
(gdb) bt full
#0 0x0044da80 in ast_var_name (var=0x10f1d58a0) at chanvars.c:69
name = 0x10f1d58b0 Address 0x10f1d58b0 out of bounds
#1 0x0049948f in pbx_builtin_setvar_helper (chan=0xf460320,
name=0x2aaabf53cbf7 DIALSTATUS,
value=0x417a0690 BUSY) at pbx.c:5825
newvariable = (struct ast_var_t *) 0x10f1d58a0
headp = (struct varshead *) 0xf460880
nametail = 0x2aaabf53cbf7 DIALSTATUS
__PRETTY_FUNCTION__ = pbx_builtin_setvar_helper
Great, this confirms that i'm not the only one having this problem. Can
you please add this to http://bugs.digium.com/view.php?id=10923
As from my knowledge - this will happen often on 1.4.13.. The safe
version i'm using is 1.4.10, but 1.4.12.1 already have this problem..
Regards,
Atis
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Re: [asterisk-users] Faxing with Asterisk SpanDSP [Was Fax Problems with SpanDSP]
On Wed, 31 Oct 2007, Alan Lord wrote: Steve Underwood wrote: snip / ... SpanDSP handles faxes within Asterisk, through the app_rxfax and app_txfax applications. It handles faxes outside Asterisk when used with iaxmodem (there is actually a copy inside the iaxmodem package). SpanDSP cannot be used by the standard distribution of Asterisk, as it is GPL code. However, if you are using Asterisk within the restrictions of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily. Regards, Steve Thanks Steve, I was a bit lazy when I posted that question sorry - I just got a bit excited. A quick google and looking around the voip-info.org site pointed me to some interesting pages. Could anyone who has experience confirm if: * this will work with an x100p card so I can have a single FXO line that can detect incoming faxes and/or voice calls and enable me to use it for outgoing voice and fax? I've used it in exactly this mode with TDM400 cards. It worked mostly OK. You might need to fiddle with the gains in the zapata.conf file. (although I now steer people away from this way of doing it and use a separate ATA if they have a real fax machine, or an external fax to email service) Also, I read somewhere that Asterisk should never be used with fax. IIRC correctly it was by the moderator on the Trixbox forum discussion about the trial of OSLEC. Yes, here: http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems I really don't know how many times I need to say this but you should never run faxes through Asterisk. It was not designed to handle it. -- Kerry Garrison trixbox Community Director Is he right? Dunno, but it works. FAX data is nothing special - however modems are very critical of jitter - you and me and tolerate a bit of packet loss, or the odd duplicate, etc. in an audio stream, a modem can't. (NO CARRIER :) So your asterisk box shouldn't have jitter internally, but if it's running cpu intensive applications, it might have, I guess... A lot of people have success with running iaxmodem - which is just spanDSP connected to some fancy code to understand AT commands, and they then plumb this into hylafax running on a separate server over ethernet.. I've tried ATAs over ethernet to send/receive faxes to real fax machines with good results too. (Make sure the codec is G711 though, as nothing else will work) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 required for IP---TDM---IP
On Tue, 30 Oct 2007, satish patel wrote: Dear all I have already post this question but i need more input for this setup [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention] Asterisk - codec (G.711/ulaw) Avaya - codec ( G.711/ulaw) Now I need G.729 on my asterisk side and i have put G.729 codec setting on my IP phone and when i make call from asterisk to Avaya Extention i got error translator not in path so i need to get license of g.729 on asterisk for transcoder or it will work wothout translator ??? My question is :-- Is there Required G.729 (License) on Asterisk Or Not ??? You can purchase them from Digium: http://store.digium.com/productview.php?category_id=5product_code=8G729CODECmain_category_id=5 $10 each. Install one license for each simultaneous g792 call you expect to take on the asterisk box and off you go. There are free versions of g729 avalable, but if your country is compatable with the various (US) patent laws then you ought to pay the license fee to stay legal. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
On 10/31/07, Douglas Garstang wrote: I guess... it shouldn't be too hard to find the time out value in the source and change it I couldn't find any timeout related parameter in app_addon_sql_mysql.c You may find a default value in one of the header files. I am wondering if it wouldn't be easier to try and detect the existence of the DB host via System(). -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI over T1 calls dropping, cause 100
The T1 was setup as tie line, not a trunk. The Bell guy tried setting up the line 2 ways: 1. As a trunk. This did not work because: a) When he typed in the access code for the trunk on a phone set (and then any numbers), the call never appeared on the Asterisk side. b) The Bell guy said that unless Asterisk was generating a dialtone, a trunk would not work. (I struggled to understand these explanations...but figured I must be missing something) 2. As a tie line. This sort of worked because a) When he typed the access code for the tie line on a phone set, he got a second dial tone. b) When he dialed any digits thereafter, the call was handed across the T1 and I saw it on the asterisk side. Can you give me any specifics (or a link) on how the Meridian side should be configured? Thanks, MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: Tuesday, October 30, 2007 11:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI over T1 calls dropping, cause 100 -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component -- Processing IE 108 (cs0, Calling Party Number) The Meridian is trying to Invoke the Remote Operations Service Element (ROSE). That is used to support interactive applications. My guess is that Meridian thinks its talking to another Meridian and its trying to startup some application. That is not going to play well with Asterisk. You need to see how to disable that, or configure a plain trunk without any fancy stuff at the Meridian side. Andres. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
On Tuesday 30 October 2007 18:19:33 Douglas Garstang wrote: Anyone know if the MySQL() application has a configurable timeout? It does not. If it tries to connect to a bogus IP, it's timeout seems to be a few minutes. I'd like to cut it down to a few seconds. The key would be adding this line at the appropriate point: mysql_options(mysql, MYSQL_OPT_CONNECT_TIMEOUT, timeout) where timeout is an integer. Remember that it needs to be set BEFORE the connection. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank + AsteriskNOW
On Wed, Oct 31, 2007 at 10:54:22AM -0200, Guilherme Loch Waltrick Góes wrote: Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in mantis seems to be closed, but I cannot find fxload or lsusb to do some debugging. Please use the latest beta: 6.5: http://www.rpath.org/rbuilder/project/asterisk/ Those issues, along with a number of smaller issues, have been resolved there. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astribank + AsteriskNOW
Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in mantis seems to be closed, but I cannot find fxload or lsusb to do some debugging. -- Guilherme Loch Góes MSN:[EMAIL PROTECTED] (48) 99115299 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct voltages but no dial tone on TDM2400P
On Tuesday 30 October 2007 18:22:21 Alex R Green wrote: The TDM2400P card has three green FXS modules close to the 50pin connector and three red FXO modules at the rear. The card was installed in the system prior to loading Trixbox. I think the card is working: zaptel.conf was set up correctly with the first 12 channels fxsks and the last 12 channels fxo kewlstart. Your signalling is backwards. The first 12 channels are fxs modules, but they are signalled with fxo signalling, so fxoks=1-12, and the last 12 are fxo modules, but they are signalled with fxs signalling, so fxsks=13-24. Ditto for zapata.conf. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] flooded by Maximum trunk data space exceeded messages
try to reduce number of calls on trunk or create multiple trunks. On 10/31/07, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hi, Using 1.4.13 and trunking a single iax channel to a similar box my asterisk console is flooded with: [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space exceeded to xx.xx.xx.xx:4569 Known issue? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues with downloads.digium.com
On a slightly different matter: http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri 1.4.1 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best cheap card to use for home Asterisk system???
Hi all - I'm building an Asterisk system (Trix2.2) for the house- I'd like to do the following things: I have a single phone line (happens to be Charter Communications VOIP, but I have their ATA and they've connected to red/green pair in the house wiring) What I'd like to do is this: Get some low-end but reliable card/external adapter which would connect to their ATA and tie into Asterisk to take calls and faxes I'm assuming this should be something with one FXO and one FXS port to connect the incoming line to and to connect the red/green wiring in the house to. I don't mind if all the house phones ring at one time for the moment, as line 2 on them are the Asterisk extensions. Whatever I use must also have failover capability, such that when Asterisk is not working right (server down completely OR just not responding) then the unit fails over and cross connects and makes things work just as is normally the case with Charter only. Unfortunately, I don't have a budget of hundreds of dollars for a true Digium multiport card - I've already built out Asterisk and have a Cisco ATA supporting line 2 on a couple of cordless phones, but I'd like to have the failover piece so that if * starts failing, the home phones still work.. Thanks, Tim ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.13 -- issue with parked calls
I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. Having an issue with (I think) parked calls. We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a specific pattern that I've discovered so far. I had this happen to me personally this morning -- receptionist parked a call for me on extension 7001, but when I dialed 7001, just got dead air. I could see in asterisk that the call was indeed parked though, and after calling the person back, he reported he was just hearing the lovely on-hold music. Is there a known issue (and even better, a fix) for this situation? Any other information I can provide I'll do so! -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
Tilghman Lesher wrote: The key would be adding this line at the appropriate point: mysql_options(mysql, MYSQL_OPT_CONNECT_TIMEOUT, timeout) where timeout is an integer. Remember that it needs to be set BEFORE the connection. Anybody like to give more detailed instructions for those of use not instructed in C? Thanks! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP phone recommendation (used to be: no subject)
I'd go with Polycom all the way. We have a number of different types of phones in use, or that we've worked with, including Grandstream, SIpura and Atacom, and the quality difference with the Polycom phones is astounding. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: My apologies to the list for not having entered a subject line in the email. Thanks On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
Find this line: if (mysql_real_connect(mysql, dbhost, dbuser... Add this before that line: int timeout = 10; /* 10 second timeout */ mysql_options(mysql, MYSQL_OPT_CONNECT_TIMEOUT, (const char *) timeout); And recompile. On 10/31/07, Doug Lytle [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: The key would be adding this line at the appropriate point: mysql_options(mysql, MYSQL_OPT_CONNECT_TIMEOUT, timeout) where timeout is an integer. Remember that it needs to be set BEFORE the connection. Anybody like to give more detailed instructions for those of use not instructed in C? Thanks! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.13 -- issue with parked calls
Barry D. Hassler wrote: Is there a known issue (and even better, a fix) for this situation? Any other information I can provide I'll do so! How are the calls getting parked? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best cheap card to use for home Asterisk system???
Tim Reimers wrote: I have a single phone line (happens to be Charter Communications VOIP, but I have their ATA and they’ve connected to red/green pair in the house wiring) Ok. so they've installed an ATA which connects your analog phones to their VoIP (perhaps SIP) service. What I’d like to do is this: Get some low-end but reliable card/external adapter which would connect to their ATA and tie into Asterisk to take calls and faxes OK. Since we've established above that Charter's service is VoIP converted to analog, AND since Asterisk isn't really designed to work with fax over IP it is safe to say that it's not worth the effort to attempt to get this to work. I have relatives who have Time Warner's offering and even a stand alone fax machine will not work reliably over their internet phone service. Hell the audio quality is crap most of the time. I’m assuming this should be something with one FXO and one FXS port to connect the incoming line to and to connect the red/green wiring in the house to. I'm not sure if you're familiar with the Canadian television show that is popular on PBS in the US, but this sounds alot like the guy on the Red Green Show using duct tape to fix things. If you really want to use Asterisk, you'd be better off getting an account with a SIP provider and using an FXS adapter to feed line 2 on your phones similar to what Charter is doing with line 1. Linksys makes a decent adapter which would suit this purpose. Good luck! Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI commands missing...
This also happens if zaptel fails to load. Check your messages file. John covici wrote: Well, this happened to me one time when I forgot to compile the pri library before the asterisk! Could you have done that? on Wednesday 10/31/2007 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Wed, Oct 31, 2007 at 12:06:25AM -0600, Carlos Chavez wrote: I have an Asterisk server running Elastix but patched to use Unicall. Everything seems to be working fine and the TE220 card is up and running with port 1 configured as PRI and port 2 as MFC/R2. We can already send and receive calls on port two but we cannot on port one. That is when we noticed that there are no PRI commands available on the Asterisk CLI. We cannot use PRI DEBUG SPAN to determine why port 1 is not receiving or sending calls. Why would this commands be missing? I wonder how those two should interact. The first thing chan_zap tries to do is to open all of its spans. Maybe it has failed there? Try playing with [trunkgroups] to explicitly tell it to only touch the Zaptel spans that are PRI. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Size of Exten when using IAX
If I look at the console (with verbosity on 3) I see that also the last 4 characters are lost. I never heard of 'wireshark on the wire' I'll try this. Is IAXVARS also supported on asterisk 1.0.0 ? -- Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: dinsdag 30 oktober 2007 15:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Size of Exten when using IAX On Tuesday 30 October 2007 08:40:51 Arjan Kroon | Mobillion wrote: We are use IAX protocol between two asterisk servers. Now we send information through this protocol by using EXTEN We see that the variable EXTEN only holds 66 characters. If we set a value larger then 66 characters, for example 70 characters. The last 4 characters are cut off. Is there a way to increase this variable? You're going to have to provide more information for us to help you. There are numerous places where the extension string could be getting truncated, so you'll have to look some more: 1) On the console, with verbose set to 3 or higher, when the dialplan is executed, are you showing all of the numbers? 2) If you run wireshark on the wire, does the IAX2 packet show all of the numbers in the CALLED_NUMBER IE? Also, you should know that in trunk, there is a much better way of transmitting independent bits of data about the call, called IAXVARS. We're presently looking at abstracting this into something a bit more protocol independent, but that's the way it is presently. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
Sean Bright wrote: Find this line: if (mysql_real_connect(mysql, dbhost, dbuser... Excellent! Thank you both! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issues with downloads.digium.com
On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote: On a slightly different matter: http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri 1.4.1 . Yes, I noticed that too and was wondering if it is just because they have not updated the site or if there is a problem with the newest versions and they do not want people to download them. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI commands missing...
On Wed, 2007-10-31 at 09:41 -0600, Anthony Francis wrote: This also happens if zaptel fails to load. Check your messages file. John covici wrote: Well, this happened to me one time when I forgot to compile the pri library before the asterisk! Could you have done that? Asterisk is working with the second span with Unicall and that would not be possible unless Libpri and Zaptel are already loaded. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mobile phone codecs ...
Not strictly asterisk related, however... Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?) Anyway, in a fit of idleness, I thought I'd see what codecs it supports, as I couldn't find it in the manual... And it supports: ilbc g729 ulaw/alaw No GSM! How odd is that, given that it's a GSM mobile phone... Anyway, my quest for the ultimate one handset solution is getting closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor Granite it might have half a chance of working outside the room with the access point, however ... Anyone tried the Plantronics Voyager 510 bluetooth headsets which regsiters to both a mobile phone and their own base unit (which presumably has a USB sound device) as in: https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click I'm not a fan of soft-phones, and not sure I want to have a borg implant on when I'm not driving, but ... Oh well... Back to the grind! Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 Faxing and Asterisk
Nasir Iqbal wrote: Hi, Have you tried Callweaver http://www.callweaver.org I was really hoping to be able to use Trixbox to do this and it's a pretty complete solution by itself. Unfortunately that requires Asterisk. It appears that there is no way to get Asterisk, or anything on the Asterisk box, to act as a T.38 endpoint. This appears to be the result of a licensing issue with SpanDSP. http://www.voip-info.org/wiki/view/T.38 That's a real shame as T.38 termination support is one of the last big pieces for us to make Asterisk a seamless solution. Paul Bryson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP_INFO
Hi list, does anyone of you know wether asterisk can handle SIP_INFO on pure sip calls? Is that something I have to handle in the extensions? Does asterisk hand incoming SIP_INFO over to an already connected peer? Thanks and regards, Christophorus Laube ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queues without 302 redirects?
Hi, Using 1.4.13 is it possible to ignore 302 redirects from sip devices belonging to a queue? For a queue that rings the whole office it doesn't seem very useful to obey a redirect programmed on a phone. It seems this was the default behaviour in 1.2. Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.13 -- issue with parked calls
Barry D. Hassler wrote: I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. Having an issue with (I think) parked calls. We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a specific pattern that I've discovered so far. I had this happen to me personally this morning -- receptionist parked a call for me on extension 7001, but when I dialed 7001, just got dead air. I could see in asterisk that the call was indeed parked though, and after calling the person back, he reported he was just hearing the lovely on-hold music. Is there a known issue (and even better, a fix) for this situation? Any other information I can provide I'll do so! What kind of phones are you using? are they Zap or SIP? Can you provide a CLI output with any tips in it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues without 302 redirects?
On Wed, Oct 31, 2007 at 06:11:47PM +0100, Louis-David Mitterrand wrote: Hi, Using 1.4.13 is it possible to ignore 302 redirects from sip devices belonging to a queue? For a queue that rings the whole office it doesn't seem very useful to obey a redirect programmed on a phone. It seems this was the default behaviour in 1.2. For the record and google the answer is the 'i' option in Queue(). Thanks again to Strom_M on #asterisk! god I love IRC... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI commands missing...
On Wed, Oct 31, 2007 at 10:46:42AM -0600, Carlos Chavez wrote: On Wed, 2007-10-31 at 09:41 -0600, Anthony Francis wrote: This also happens if zaptel fails to load. Check your messages file. John covici wrote: Well, this happened to me one time when I forgot to compile the pri library before the asterisk! Could you have done that? Asterisk is working with the second span with Unicall and that would not be possible unless Libpri and Zaptel are already loaded. chan_unicall uses zaptel (the kernel interface) but not chan_zap.so (and not libpri, IIRC). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL() timeout
On 10/31/07, Doug Lytle wrote: Excellent! Thank you both! Doug don't forget that line of code will disappear the next time you upgrade your * addons, unless the change makes it into the official code base. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have used the Grandstream GPX2000, HT503 and GXW4104 gateways. Quality is in all cases are on the lower end. The quality I refer to is buggy software and poor call quality. I have been involved with Telecom since the early 80s and dealt with a lot of phone systems. The Grandstream phones just plain feel cheap. Real Walmart quality, not professional business class equipment. The phone functioned ok and was super easy to setup but complaints of echo and poor volume levels were common. They may be better as we have not used them in over 6 months. We have recently used their gateways due to good pricing and their economics fit our solution base well but ran into issues with them. I believe their gateways will get improved as both are new and on early firmware releases. However, we got upset with poor support. Either no call back at all or a useless email a day later with little to no information to help solve our issue. In Grandstream's defense it may be we are just too small to matter and that's ok. We prefer to go elsewhere and deliver product that when the average user picks it up to talk on it they say this is quality stuff. Asterisk is as talented as the firm that programs it BUT the phone is crucial in the end user's system satisfaction. Regardless of what you put in the back room the phone IS the device that sets the impression to your client if you are delivering a quality solution. We would do Cisco because it is high quality but we don't care to fight with the configuration or licensing issues. We would do Polycom, and probably will, but have not had the time to jump to through the hoops needed to acquire good enough pricing to make money selling them. We feel Aastra is a good compromise in delivering quality product to make the customer happy with their decision while still making us to make some sort of small profit for our time. It's solid and provides a quality feel and function. This said, Grandstream is not junk and this is not meant to be a Grandstream rant. I would like to apologize if I jumped in too quick sounding that way. Grandstream is just the lower end of quality and should be deployed in applications where the client is willing to accept that. That's not our marketplace. If you want easy to configure, low cost, slam dunk Asterisk deployments then Grandstream works. But the end result will not be as good if you build a system with Cisco, Polycom, Snom, or Aastra. We've even tested Avaya 46XX phones on Asterisk. They sound GREAT! Probably one of the best. We just can't get Asterisk to light the messaging waiting light on the phone. Arrggg! You need to decide what your marketplace offering is and what your clients are willing to accept. If call quality is the most important then our testing shows nobody beats Polycom or Avaya. Someday we are going to beat the Avaya message waiting light issue. If quality of deskset feel is the most important factor them Avaya and Cisco stand out as best. We will not put configuration into a factor simply because the customer uses the tool we are expected to configure it to their needs. We won't sell them any device based on it being easier for us to configure. I would like to hear what people say about Snom as their sets look very nice. Sorry for the novel, all I really wanted to express is Grandstream is cheap, look before you jump. Good luck on your decision... Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in
Re: [asterisk-users] PRI over T1 calls dropping, cause 100
On 10/31/07, Michelle Dupuis [EMAIL PROTECTED] wrote: The T1 was setup as tie line, not a trunk. The Bell guy tried setting up the line 2 ways: 1. As a trunk. This did not work because: a) When he typed in the access code for the trunk on a phone set (and then any numbers), the call never appeared on the Asterisk side. b) The Bell guy said that unless Asterisk was generating a dialtone, a trunk would not work (I struggled to understand these explanations...but figured I must be missing something) There is no dialtone on a PRI/T1. I think what he meant was you need to change in your zaptel config pri_cpe to be pri_net then it will allow you to setup that trunk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank + AsteriskNOW
You will need to go to rPath www.rpath.org and download AsteriskNOW version 6.5 beta. All should work immediately. Rupert Utteridge Tel:+61 2 9037 4191 Message: 18 Date: Wed, 31 Oct 2007 10:54:22 -0200 From: Guilherme Loch Waltrick G?es [EMAIL PROTECTED] Subject: [asterisk-users] Astribank + AsteriskNOW To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in mantis seems to be closed, but I cannot find fxload or lsusb to do some debugging. -- Guilherme Loch G?es MSN:[EMAIL PROTECTED] (48) 99115299 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071031/7e4ea9 14/attachment-0001.htm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile phone codecs ...
On 10/31/07, Gordon Henderson [EMAIL PROTECTED] wrote: Not strictly asterisk related, however... Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?) Anyway, in a fit of idleness, I thought I'd see what codecs it supports, as I couldn't find it in the manual... And it supports: ilbc g729 ulaw/alaw No GSM! How odd is that, given that it's a GSM mobile phone... Anyway, my quest for the ultimate one handset solution is getting closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor Granite it might have half a chance of working outside the room with the access point, however ... Anyone tried the Plantronics Voyager 510 bluetooth headsets which regsiters to both a mobile phone and their own base unit (which presumably has a USB sound device) as in: https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click I'm not a fan of soft-phones, and not sure I want to have a borg implant on when I'm not driving, but ... Oh well... Back to the grind! Gordon ___ I think that's pointless. Why do you need a USB audio device? You can pair it to the computer directly and use it with any soft phone. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mark Spencer on Pulver TV
May be of interest to you: http://www.blogtv.com/Shows/96/YeTrZe3vb2Vpos=ancr ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with Asterisk SpanDSP [Was Fax Problems with SpanDSP]
Steve Underwood wrote: SpanDSP cannot be used by the standard distribution of Asterisk, as it is GPL code. However, if you are using Asterisk within the restrictions of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily. I was wondering how someone could modify Asterisk to be GPL compliant? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Long duration calls with Asterisk out to VoIP telco
Hi list, My long duration calls are being timeout by my SIP VoIP provider for failure of receiving re-INVITE within their timeout limit. Is there a way to config Asterisk to automatically send a re-INVITE message every 10 to 15 minutes? I looked into the sip.conf file and couldn't find such a parameter. Thnx. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Honestly, Its my opinion that the Aastra phones are very lacking in the firmware department. If they could get that sorted out I wouldn't mind using them. But for now there are too many NAT issues mostly caused because they use an OLD version of Broadcom CallCtrl. Why they use an ancient version is beyond me but the phones dont even have a NAT keepalive option. They promise updates to their firmware but then they only fix minor bugs. Grandstream are ok. But as others have said their support is very lacking. I've had products of theirs behave very oddly like operate and refuse to apply any settings no matter what and not allow a factory reset... paperweight. I'd personally use Polycom in the situations where there's no NAT and the Linksys SPA-phones where you do have NAT. On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with Asterisk SpanDSP [Was Fax Problems with SpanDSP]
[EMAIL PROTECTED] wrote: Steve Underwood wrote: SpanDSP cannot be used by the standard distribution of Asterisk, as it is GPL code. However, if you are using Asterisk within the restrictions of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily. I was wondering how someone could modify Asterisk to be GPL compliant? It *is*. The issue is that since Asterisk is dual-licensed, GPL-only code cannot be included in the mainline asterisk source. SpanDSP is (to my knowledge) GPL-only and hence, cannot be included. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 Faxing and Asterisk
I thought there was some talk of getting T38Gateway into asterisk_addons? Stupid linking bullshits. On 10/31/07, Paul Bryson [EMAIL PROTECTED] wrote: Nasir Iqbal wrote: Hi, Have you tried Callweaver http://www.callweaver.org I was really hoping to be able to use Trixbox to do this and it's a pretty complete solution by itself. Unfortunately that requires Asterisk. It appears that there is no way to get Asterisk, or anything on the Asterisk box, to act as a T.38 endpoint. This appears to be the result of a licensing issue with SpanDSP. http://www.voip-info.org/wiki/view/T.38 That's a real shame as T.38 termination support is one of the last big pieces for us to make Asterisk a seamless solution. Paul Bryson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 required for IP---TDM---IP
Here's a link to the free version: http://asterisk.hosting.lv/ On 10/31/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Tue, 30 Oct 2007, satish patel wrote: Dear all I have already post this question but i need more input for this setup [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention] Asterisk - codec (G.711/ulaw) Avaya - codec ( G.711/ulaw) Now I need G.729 on my asterisk side and i have put G.729 codec setting on my IP phone and when i make call from asterisk to Avaya Extention i got error translator not in path so i need to get license of g.729 on asterisk for transcoder or it will work wothout translator ??? My question is :-- Is there Required G.729 (License) on Asterisk Or Not ??? You can purchase them from Digium: http://store.digium.com/productview.php?category_id=5product_code=8G729CODECmain_category_id=5 $10 each. Install one license for each simultaneous g792 call you expect to take on the asterisk box and off you go. There are free versions of g729 avalable, but if your country is compatable with the various (US) patent laws then you ought to pay the license fee to stay legal. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large voicemail
Probably the best option is store the messages in IMAP and the userdate in a database. Honestly I dont think there is an issue with any number of mailboxes the issue is going to be how many calls at once your system can handle or how well your architecture scales to handle multiple machines. Can your storage handle 5,000 mails being recorded at once? Just trying to sort out the thousand different aspects of it all in my mind right now I say you give it a try but expect to write your own voicemail fron the ground up and not necessarily based on Asterisk. Then again, I could be wrong. On 10/25/07, Pepo [EMAIL PROTECTED] wrote: I am trying to use Asterisk as the voicemail system of the TELCO where I work. I wanna test with 2 mail boxes ( and later with a better machine/server I hope try with 7 ). How do I include in voicemail.conf the file with the mail boxes?, In a big system like this,is better use text files or any database? Thanks -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile phone codecs ...
Am Mittwoch, den 31.10.2007, 16:47 + schrieb Gordon Henderson: Not strictly asterisk related, however... No GSM! How odd is that, given that it's a GSM mobile phone... Maybe the GSM codec is implanted to the GSM chip and that one does alaw, ulaw... Anyway, my quest for the ultimate one handset solution is getting closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor Granite it might have half a chance of working outside the room with the access point, however ... That is one of the two points I like at the American style wood and wallpaper houses (the other being that construction is cheap and easy, in comparison). Living in a concrete house is not all the best thing as well though. My Pirelli dual-mode phone loses WLAN link just outside my flat door in the hall way, one concrete wall and about five meters from the Access Point. What luck they only used drywall inside the appartment. Anyone tried the Plantronics Voyager 510 bluetooth headsets which regsiters to both a mobile phone and their own base unit (which presumably has a USB sound device) I had a Plantronics device here that connected to a phone-line-tap base station or to my mobile via bluetooth. I did not buy it though because it only worked with my Sony T610 (stone-age old, about 2003), not with my O2 xda. I sold one plantronics 510 to a customer who uses it with his Nokia Esomething, and really likes it. AFAIK the USB device that comes with it is a bluetooth dongle, not a virtual audio device, but that might be different between versions of that device, and my customer definitely does not use it. I noticed with my plantronics device back then that you needed to re-pair it (whohoo, never noticed that similarity repair to re-pair) to whatever device you want to use it with, and that sucked because it took half a minute and some interaction with the mobile or base station. as in: https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click I'm not a fan of soft-phones, and not sure I want to have a borg implant on when I'm not driving, but ... resistance is futile :-# Best regards, Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk configuration for T1 CAS lines
Hi, I am trying to use Asterisk PBX with T1 CAS. The setup that I am looking for is as below Analog phones == Asterisk T1 CAS === Integrated Access Device IP Network for VoIP. The Asterisk has a T1 card and I want a CAS config between Asterisk and T1 port of IAD. The Asterisk has got a FXS card to which the analog phones are connected. I would like to know whether T1 CAS configuration is possible with Asterisk. If possible, any pointers to configuration would be really helpful. Thanks and Regards, Jana ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing with Asterisk SpanDSP [Was Fax Problems with SpanDSP]
On Wednesday 31 October 2007 13:55:32 [EMAIL PROTECTED] wrote: Steve Underwood wrote: SpanDSP cannot be used by the standard distribution of Asterisk, as it is GPL code. However, if you are using Asterisk within the restrictions of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily. I was wondering how someone could modify Asterisk to be GPL compliant? Asterisk is already GPL compliant. The issue is that the code has exceptions to the GPL, to allow it to be linked to certain patented codecs, for example, without violating patent licenses or the GPL. SpanDSP, on the other hand, is not dual licensed, so you cannot run, for example, both G.729 and SpanDSP in the same binary (legally), since the pure GPL license of SpanDSP pollutes the Asterisk binary to require the same of the G.729 codec (which is not permissible until the patents on G.729 expire). -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have Cisco 9760 for executives and Aastra 9112i for everybody else. We started with Grandstream, don't remember the model, cost around $80 USD but it had bad audio quality and echo problems (running asterisk 1.09). The quality of construction felt poor, like a toy phone. We replaced them with the Aastra for double the cost and the quality improved dramatically. Audio quality was much better and echo problems all but eliminated. This phone also feels more solid. There are a few areas that are not perfect; the speaker phone is good not excellent and we have had to replace a couple of phones because they have stopped working. Over all I would say not bad for the price especially if they are for general use. We had to upgrade from the Aastra phones for our executives because they needed very good audio for both handset and speaker phone. We are using Cisco 9760's for them and have had no problems with quality. Plus they have a very solid feel. My question to the list is: As I need to add phones I am considering buying used Cisco 9760's. Is there any difference with the 9760G? I have heard that the 9761's have even better audio quality. Our main requirement is audio quality, our users do not need a lot of features on their phones. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Wednesday, October 31, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with flash hook
Hi,
I facing a problem with flash hook. When ever I do a flash hook to place an
extsing call on hold, the call gets disconnected. The debugs on Asterisk
shows that 'on hook event detected' when I press the flash button on the
phone. The setup is like this
Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD
and configured for ISDN PRI lines. Analog phones come out of the FXS cards.
Phone 1
Phone 2 --FXS card . Asterisk. T1 card ISDN PRI=IAD
+IP
Phone 3
The Asterisk is configured to plce calls on one of the T1 ports/channels
based on the dial plan. ie, if I dial 94XX then the call will be
placed on a channel in the 4th T1 port which is connected to the IAD. I am
able to make calls using this setup. But I am not able to put the call on
hold using a flash hook. The call gets disconnected when the flash button is
pressed.
I have transfer=yes, threewaycalling=yes etc enabled in my zapata.conf .
signalling=fxo_ks
threewaycalling=yes
transfer=yes
callwaiting=yes
flash=1000
context=from-internal
group=1
callgroup=1
pickupgroup=1
hidecallerid=no
usercallerid=yes
musiconhold=default
channel = 97
Here are the asterisk debugs that I get when I flash the call
-
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Exception on 106, channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Got event On hook(1) on channel 97
(index 0)
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Enabled echo cancellation on
channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Echo cancellation already on
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Unlinking slave 73 from 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Removed 83 from conference 9/97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Removed 106 from conference 9/73
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Updated conferencing on 97, with 0
conference users
Oct 25 10:27:13 DEBUG[28965] channel.c: Returning from native bridge,
channels: Zap/97-1, Zap/73-1
Oct 25 10:27:13 DEBUG[28965] channel.c: Hanging up channel 'Zap/73-1'
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: zt_hangup(Zap/73-1)
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option AUDIO MODE, value: ON(1)
on Zap/73-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Hangup: channel: 73 index = 0,
normal = 83, callwait = -1, thirdcall = -1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Not yet hungup... Calling hangup
once with icause, and clearing call
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 73
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/73-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Updated conferencing on 73, with 0
conference users
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/73-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 73
Oct 25 10:27:13 VERBOSE[28965] logger.c: -- Hungup 'Zap/73-1'
Oct 25 10:27:13 DEBUG[28965] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Oct 25 10:27:13 DEBUG[28965] app_macro.c: Spawn extension
(macro-dialout-trunk,s,14) exited non-zero on 'Zap/97-1' in macro
'dialout-trunk'
Oct 25 10:27:13 DEBUG[28965] pbx.c: Spawn extension
(macro-dialout-trunk,s,14) exited non-zero on 'Zap/97-1'
Oct 25 10:27:13 DEBUG[28974] app_queue.c: Device 'Zap/73' changed to state
'0' (Unknown) but we don't care because they're not a member of any queue.
Oct 25 10:27:13 DEBUG[28965] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Oct 25 10:27:13 DEBUG[28965] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-10-25
10:26:53','6782363001','6782363001','946782362001','from-internal',
'Zap/97-1','Zap/73-1','Dial','ZAP/g4/6782362001',20,17,'ANSWERED',3,'','
1193322403.4539')
Oct 25 10:27:13 DEBUG[28965] channel.c: Hanging up channel 'Zap/97-1'
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: zt_hangup(Zap/97-1)
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Hangup: channel: 97 index = 0,
normal = 106, callwait = -1, thirdcall = -1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/97-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Updated conferencing on 97, with 0
conference users
Oct 25 10:27:13 VERBOSE[28965] logger.c: -- Hungup 'Zap/97-1'
Oct 25 10:27:13 DEBUG[28976] app_queue.c: Device 'Zap/97' changed to state
'0' (Unknown) but we don't care because they're not a member of any queue.
Oct 25 10:27:14 DEBUG[3779] chan_zap.c: Monitor doohicky got event Hook
Transition Complete on channel 97
Oct 25 10:27:14 DEBUG[3779] chan_zap.c: Monitor doohicky got event
Ring/Answered on channel 97
Oct 25 10:27:14 WARNING[3779] chan_zap.c: zt hook
[asterisk-users] h323 help
We've configured ooh323 on our 1.4.6 asterisk server. We've looked at various sites for tips, most recently http://www.tek-tips.com/viewthread.cfm?qid=1243330page=3. The module seems to load properly. When we do a tcpdump, we see traffic flowing between the asterisk server and the Avaya communication manager. However, we're not geting phone calls connect. Since we do not manage the Avaya CM, how can we further verify that our ooh323 config is correct? Thanks for any tips. -- Jiann-Ming Su I have to decide between two equally frightening options. If I wanted to do that, I'd vote. --Duckman The system's broke, Hank. The election baby has peed in the bath water. You got to throw 'em both out. --Dale Gribble Those who vote decide nothing. Those who count the votes decide everything. --Joseph Stalin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 Faxing and Asterisk
[EMAIL PROTECTED] wrote: I thought there was some talk of getting T38Gateway into asterisk_addons? Stupid linking bullshits. Stupid indeed. I'm surprised T.38 support isn't a higher priority for Digium, given that faxing has such a high failure rate with VoIP. Paul ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer on Pulver TV
On 10/31/07, randulo wrote: May be of interest to you: http://www.blogtv.com/Shows/96/YeTrZe3vb2Vpos=ancr not long enough ! Hot tip : you can skip the first 6 minutes thnx ! -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 and Callbacks
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
---
context LegA {
_X. = {
Dial(SIP/[EMAIL PROTECTED]);
}
}
And my default context:
--
context default {
callback = {
NoCDR();
Wait(1);
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
}
}
The A leg is established, and once Asterisk goes to dial the B leg...
-- Executing [EMAIL PROTECTED]:1] Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/Provider-09a8cff8 is making progress passing it to Local/[EMAIL
PROTECTED],2
-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
== Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed so falling
back to exten 's'
== Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed so falling
back to context 'default'
[Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel
'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context
'default', but no invalid handler
Uhm, why? I have a default context with a callback extension. Of course I have
no explicit priority 1 though... this is AEL2
What's it complaining for?
Doug.
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[asterisk-users] Druid
Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Failed
After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],,r) exten = _NX,n,Hangup() ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hostname in MySQL CDR records
I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? Thanks... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax: they hear me, but I don't hear them...iptables?
I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for '[EMAIL PROTECTED]' with cause 'facility rejected' 11:03:35 Rejected registration for 'home@my-office-ip-address' with cause 'facility rejected' and on the asterisk server at work I get: NOTICE[5072]: chan_iax2.c:5252 register_verify: No registration for peer 'home' (from my-home-ip-address) iax.conf: [general] jitterbuffer=yes tos=ef iaxcompat=yes [iax-from-home] type=user allow=gsm username=home secret=password context=long-distance ;;auth=md5,plaintext [iax-to-home] type=peer allow=all username=home secret=password ;;auth=md5,plaintext [guest] type=user context=long-distance callerid=Guest IAX User This seems like a NAT problem, but how can that be with iax if they can hear me? Doesn't everything travel on the same port? How does my phone ring? Iny any event, my home iptables -L: Chain FORWARD (policy ACCEPT) target prot opt source destination ACCEPT all -- 192.168.2.0/24 anywhere ACCEPT all -- 10.0.0.0/8 anywhere ACCEPT udp -- anywhere anywhereudp dpts:1024:65535 ACCEPT udp -- anywhere anywhereudp dpt:avt-profile-1 ACCEPT udp -- anywhere anywhereudp dpt:sip . Any help appreciated sean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.13 -- issue with parked calls
We park the calls by transferring to extension 7000, which is our parking extension. We have both Zap and SIP extensions, and I haven't been able to see a pattern if its related to one or the other. The primary person answering the phone is using a SIP phone (Grandstream GXP-2000), we have a small number of analog phones left (2), and other SIP phones (mostly Polycom). The only clue I've seen with the CLI is that I'll generally see a LOT of entries for active channels on extension 7000 if I do a show channels. I'll try to catch this situation again and grab the output. I only started having this problem when I upgraded to the 1.4 version from 1.2. On 10/31/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Barry D. Hassler wrote: I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. Having an issue with (I think) parked calls. We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a specific pattern that I've discovered so far. I had this happen to me personally this morning -- receptionist parked a call for me on extension 7001, but when I dialed 7001, just got dead air. I could see in asterisk that the call was indeed parked though, and after calling the person back, he reported he was just hearing the lovely on-hold music. Is there a known issue (and even better, a fix) for this situation? Any other information I can provide I'll do so! What kind of phones are you using? are they Zap or SIP? Can you provide a CLI output with any tips in it? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 and Callbacks
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What do you get if you do dialplan show default? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHKUlTDQNt8rg0Kp4RAteuAJ9kbLC77Bw7G789uOIaQ1hR+++87gCgqNPB p4jMkvOg6kuVylFKaHLPwAs= =ajMg -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connection astrisk to a RAS (portmaster)
Here's my planed setup :
PRI from telco -- (port 1 of A104d) * (port 2 of A104d) -- PM3
The PM3, for those who don't know is lucent's portmaster RAS dial-up router.
I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards).
In wancfg, I have port 1 as TDM_VOICE, with hardware echo on, span 1. Port 2
is TDM_VOICE, without hw echo, Clock as master, reference 0 (for the time
being, I'm still not hooked up with my pri, will be 1 in the future), span
2. I alswo had to enable High Impedance on port 2 to operate without alarms.
Zaptel.conf:
loadzone=us
defaultzone=us
#Sangoma A104 port 1 [slot:12 bus:0 span: 1]
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#Sangoma A104 port 2 [slot:12 bus:0 span: 2]
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
zapata.conf (part of) :
;Sangoma A104 port 1 [slot:12 bus:0 span: 1]
switchtype=national
pridialplan=unknown
context=demo
group=1
signalling=pri_cpe
channel = 1-23
;Sangoma A104 port 2 [slot:12 bus:0 span: 2]
switchtype=national
pridialplan=unknown
context=demo
group=2
signalling=pri_net
channel = 25-47
Now, after start wanpipe and asterisk, my PM3 shows that the line is up (via
a t1 cross-over cable). I see that the channels are idle and waiting.
On * console, I get :
Primary D-Channel on span 2 up
All the time
I also get sometime :
== Primary D-Channel on span 2 down
[Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No
D-channels available! Using Primary channel 48 as D-channel anyway!
== Primary D-Channel on span 2 up
[Oct 31 20:51:05] ERROR[10250]: chan_zap.c:8178 zt_pri_error: !! Got reject
for frame 1, but we only have others!
In my extensions.conf, I have :
exten = 1234567,1,Dial(Zap/g2/${EXTEN})
Whem I trie that extension via a softphone, I hear hald a ring, and nothing
else. The d-channel up continue to appear on the console.
Any help would be appriciated.
Nicolas
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Re: [asterisk-users] hostname in MySQL CDR records
On Wednesday 31 October 2007 19:49:24 Jim Gottlieb wrote: I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? If you set systemname in asterisk.conf, that prefix will become part of the uniqueid field. You'll probably need to widen that field, though. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hostname in MySQL CDR records
On Wed, 31 Oct 2007, Jim Gottlieb wrote:
I would like to send the CDR records from all our machines around the
world to a single database. But I need the hostname included with each
record for monitoring purposes.
Is there a better way than using the userfield and adding
SetCDRUserfield for every call to set the userfield to the name of the
host?
Personally, I think the userfield is a hack. I prefer to add properly
named columns to the cdrs table using cdr_addon_mysql. It makes everything
so much more obvious -- especially when you don't have to cram several
values into the singularly obtuse userfield.
I prefer to retrieve the CDRs rather than send them. This way, you
only have a single script to retrieve all of the remote CDRs and the
script is simpler since you don't have to poll a directory and try to
figure out if the remote transfer has finished so you don't process a
partial file. It also makes it easier to handle a remote host that is
temporarily unavailable.
I retrieve the CDRs from remote hosts using a script that looks like this
snippet:
# for each host
for HOST in ${HOST_LIST}
do
# mark the records to be exported
mysql\
${USER_AUTH}\
--database=mumble\
--execute=update cdrs set disposition = 'EXPORTING'\
--host=${HOST}
# dump the cdrs
mysqldump\
${USER_AUTH}\
--host=${HOST}\
--no-create-info\
--skip-opt\
--where=disposition = 'EXPORTING'\
mumble\
cdrs\
/tmp/${HOST}.sql
# load the cdrs into our database
mysql\
${USER_AUTH}\
--database=mumble\
--host=localhost\
/tmp/${HOST}.sql
# delete the exported records
mysql ${USER_AUTH}\
--database=mumble\
--execute=delete from cdrs where disposition =
'EXPORTING'\
--host=${HOST}
# end of hosts loop
done
I am misusing the existing disposition column, but I never use it in my
application anyway :)
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] G.729 required for IP---TDM---IP
Thanks But is there any voice qulity effect if u use free version G.729 codec or license codec ??? If u go for license then any effect on voice qulity [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Here's a link to the free version: http://asterisk.hosting.lv/ On 10/31/07, Gordon Henderson wrote: On Tue, 30 Oct 2007, satish patel wrote: Dear all I have already post this question but i need more input for this setup [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention] Asterisk - codec (G.711/ulaw) Avaya - codec ( G.711/ulaw) Now I need G.729 on my asterisk side and i have put G.729 codec setting on my IP phone and when i make call from asterisk to Avaya Extention i got error translator not in path so i need to get license of g.729 on asterisk for transcoder or it will work wothout translator ??? My question is :-- Is there Required G.729 (License) on Asterisk Or Not ??? You can purchase them from Digium: http://store.digium.com/productview.php?category_id=5product_code=8G729CODECmain_category_id=5 $10 each. Install one license for each simultaneous g792 call you expect to take on the asterisk box and off you go. There are free versions of g729 avalable, but if your country is compatable with the various (US) patent laws then you ought to pay the license fee to stay legal. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
