Re: [asterisk-users] Portech MV3770 Caller-ID
On Thu, Mar 12, 2009 at 09:53:48AM +0100, wrote:
On 3/11/2009, Hĺkan Källberg h...@simulina.se wrote:
On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
2009/3/11 Hĺkan Källberg h...@simulina.se
Does anyone of you have Caller Presentation working in the other
direction?? My mv370 is working well, execpt the Caller ID on outgoing
GSM calls. This works with the SIM card/Provider I am using if I put
All I had to do is to enable the Caller ID ind the Mobile-Settings dialog
for each SIM (something like presentation/revocation afair). I did NOT set
the GSM number anywhere nor do I send it from Asterisk.
That is what I'd expect too, but, no...
Mobile-Settings-CLID Presentation- Supression or Invocation
it makes no difference. (and yes - I do reboot:-) When I move the SIM
to a phone, it works well...
I'm not sure wether it's an operator-specific setting or not. But i
don't handle callerid supression/invocation with the MV370, i rather do
it with asterisk.
Try simply prefix the number with *31# for invocation and #31# for
supression. Example:
[...]
exten = _06[237]0NXX!,n,Dial(SIP/*31#${ext...@gsmgw)
[...]
Again: this method may be country or even operator specific, check with
the provider if the above works. If it does, simply use the prefixes
above and forget about the MV370's settings.
It worked like a charm!! Szabó András szu...@gmail.com came up with the
same solution too. Thank you *very* much, both of you!!
Håkan
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Re: [asterisk-users] Portech MV3770 Caller-ID
On Tue, Mar 10, 2009 at 02:11:58PM +0100, Christian Victor wrote: 2009/3/10 Sasa s...@shoponweb.it Hi, I have modified in Mobile/Setting the parameter SIP From from tel/user to tel/tel and now I view the correct incoming number. Thanks. Glad I could help. It took me nearly a month to figure that out. ;-) Hello! Does anyone of you have Caller Presentation working in the other direction?? My mv370 is working well, execpt the Caller ID on outgoing GSM calls. This works with the SIM card/Provider I am using if I put the SIM card in a telephone, but not in mv370. I have tried options on the mobile setting page you talked about, with nor difference. I have tried to put the nummber of the SIM card as the Caller ID in the original Asterisk call too, just in case, but no. All I want, or rather all that would be possible is to show the number of the SIM card. Then I know that I get a transferred call from my Asterisk. It would be wonderull to get the number of the original caller too, but this will not be allowed by the provider, and this is really a Good Thing too. Håkan pgpUX8PMN2ftJ.pgp Description: 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
Re: [asterisk-users] Portech MV3770 Caller-ID
On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote: 2009/3/11 Håkan Källberg h...@simulina.se Does anyone of you have Caller Presentation working in the other direction?? My mv370 is working well, execpt the Caller ID on outgoing GSM calls. This works with the SIM card/Provider I am using if I put the SIM card in a telephone, but not in mv370. I have tried options on the mobile setting page you talked about, with nor difference. I have tried to put the nummber of the SIM card as the Caller ID in the original Asterisk call too, just in case, but no. All I had to do is to enable the Caller ID ind the Mobile-Settings dialog for each SIM (something like presentation/revocation afair). I did NOT set the GSM number anywhere nor do I send it from Asterisk. That is what I'd expect too, but, no... Mobile-Settings-CLID Presentation- Supression or Invocation it makes no difference. (and yes - I do reboot:-) When I move the SIM to a phone, it works well... I have the latest firmware: Firmware Version:Fri Sep 5 09:02:30 2008 And by the why a current 1.4 Asterisk, just updatdet to .23.2. Viele Grüße:Håkan pgpoJRRjXnEBy.pgp Description: 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
Re: [asterisk-users] Sending / Receiving sms messages with Portech 370
On Thu, Nov 20, 2008 at 05:33:34PM +, Julian Lyndon-Smith wrote: Managed to get the portech 370 up and running with asterisk (even got the callerid working!), but was wondering how (if) it is possible to send / receive sms messages through the device . All I could find googling was people asking how ;( How did you get CallerID to work?? My Portech 370 is up and running for a good while now, but without callerid... Does anyone have sms working with this device ? Well, sending from the web interface works, but utf-8 or ISO characters doesn't work. Håkan pgphMmBMpJo5o.pgp Description: 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
Re: [asterisk-users] How to capture destination number when receive call through ZAP
On Mon, Mar 24, 2008 at 10:09:34AM -0800, Mojo with Horan Company, LLC wrote:
Distinctive Ringing might be available from your telecom provider.
mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel
exten = s, n, Verbose(1|destination to ${EXTEN} )
${EXTEN} returns 's' instead of the actual destination number. Since
I have multiple phone numbers, I want to be able to route different
calls to different places.
Disregarding the fact, that the information might not be
available at all - The general trick to get it ( if available )
is to write something like:
exten = _X.,1,NoOp(The called number was: ${EXTEN})
That is: A Pattern _X. instead of s.
It works well with any Zap PRI, or mISDN line I have set up. With
analog lines I don't know.
Regards:Håkan
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Re: [asterisk-users] DECT SIP phones
On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). Hello! I would reccomend the Kirk DECT gateway. It is SIP capable and avilable for N America. We have a setup with the Skinny ( chan_sccp ) protocol and in Sweden, but I wouldn't expect any problems in NA. Our customer have used it for a while now. Regards:Håkan pgpwx3EBrFqVG.pgp Description: PGP signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Re: Spandsp rxfax txtax fails no errors
On Tue, Nov 21, 2006 at 03:33:57PM -0800, daveasterisk wrote: Is there anyone who can help with this? rxfax and txfax when called in the extensions do nothing and no error are generated that I can find. I asked something similar on the list a while ago, got no answers and took a look at the code myself and learned a little bit. Before I used System(tiff2pdf) to detect errors, which wasn't so elegant, but worked well anyway. The description looks like this: *CLI show application RxFAX -= Info about application 'RxFAX' =- [Synopsis] Receive a FAX to a file [Description] RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the given filename. If the file exists it will be overwritten. The file should be in TIFF/F format. The caller option makes the application behave as a calling machine, rather than the answering machine. The default behaviour is to behave as an answering machine. Uses LOCALSTATIONID to identify itself to the remote end. LOCALHEADERINFO to generate a header line on each page. Sets REMOTESTATIONID to the sender CSID. FAXPAGES to the number of pages received. FAXBITRATE to the transmition rate. FAXRESOLUTION to the resolution. Returns -1 when the user hangs up. Returns 0 otherwise. If you read the code, a return value of -1 means error and 0 means success, although not clearly stated so in the message above. So far, that is what you would expect, but return values are not testable in * dial plans, as far as I know. I modified app_rxfax.c to set FAXSTATUS to ERROR or SUCCESS and got it working, but then I discovered that the four return variables listed above are set only on success. I think that FAXPAGES would be the best to use for error checking. But still, you will not get a reason for the failure... There is a line in the code: ast_log(LOG_DEBUG, Fax receive not successful - result (%d) %s.\n, result, t30_completion_code_to_str(result)); that shows us that written information on the type of error *is* available. These message are in the spandsp code I suppose. Regards: Håkan pgp99onlNRg5J.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RxFAX - How to catch errors in the dialplan
Hello! We use RxFAX, version 0.0.2-pre26 over ISDN and it mostly works well. Occasionally we get .tiff files that are not convertible with tiff2pdf. This may occur if the connection is interrupted, I guess, but we also saw cases where the sender seems to have got an OK but the .tiff is not readable anyway. Therefor I'd like to check the return of the rxfax command a bit closer, but there is no return value variable. The application is said to return 0 or -1. The -1 should mean that the user hanged up. Which user, sender or receiver? What was the reason of the hangup? Error or success? And what about 0? Error or success? I am confused! Does anyone have a working dialplan to catch error/success from RxFAX? If an error occurred I don't have to worry about corrupted .tiffs. A return STATUS variable would be most welcome! I saw one dialplan example with RxFAX and priority jumping, where on jump to n+100 a retry, goto(n) was made. Does that make sense? I have seen no indication that RxFAX should support priority jumping or a j option. We use * 1.2.13. What about the 0.0.3-preX series? It was always said to be experimental and the 0.0.2 more for production use. Is this still valid, or should I try the newer 0.0.3 versions? We use current mISDN/chan_misdn, * 1.2.13, Linux kernel 2.6.13. Mostly RxFAX wokrs very well, and the users are happy with it! Best Regards: Håkan pgprpTGFWJLw3.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 320, Queues and Transfer not working as expected with * 1.2.12.1
Hello all! I have a few problems with Snom 320 phones: Problem A - Transfer out of Queues: We have a call center with some Snoms. We are using Queue and AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer a call out of the queue using the hold and transfer buttons on the Snom. This might have been the wrong way to do it all the time I found out later, but it worked. Now we upgraded to 1.12.1 and the ability to transfer out of the queue went away. Something must have changed. I tried to implement the call transfer mechanism in features.conf. I selected #1 for blind transfer and #2 for attended transfer. The * on the Snoms has an internal use. This seems to work well with other phones than the Snoms. On a Snome one have to press the check button to send the #2. This seems to work occasionally, but not reliably. The transfer call doesn't seem to work at all. I have to confess, that I only have user reports on this. Does anyone have a tip how to get this to work? The users would love to have the old functionality back - to use the specific Snom keys directly. But if we get #2 or something similar to work well, its good enough. Btw, transfer using the normal Snom buttons work well for regular calls, just not in Queues. Problem B - Quick Dial Buttons: I have used the programmable function keys together with the hint system in * to monitor local lines. It works very well, impressive! But people like to use these buttons as quick dial buttons for external numbers too. They program the buttons the same as the internal lines with hints using the Destination feature for the button. This seems to be wrong, as * gets upset over not having a hint for these requests. Trying to read the Snom manual I don't get a clear answer how to program this. Or I don't read well enough... Does anyone have a tip how to solve this?? I could set up a hint for all external numbers people like to program, but it does not scale well and would not be very meaningful. Problem C - A not Snom related transfer problem: When I use #2 to transfer a call with *s internal feature system, I need a way to go back and force between the callee and the goal of the transfer. I can't find a way to do this, either documented or elsewhere. Anyone a tip??? Thanks for your effort! Regards:Håkan pgp5qixJqNoG6.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
Hello all! On Thu, Mar 02, 2006 at 03:12:05PM -0600, Joseph Tanner wrote: The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. Joseph Tanner I see exactly this behavior too. It occurs on a system using Queue and AgentCallback Login. I have filed a bug report on this - 0006626. In this state it is not longer possible to put a call to a queue, but it is possible to place other calls through *. So * is not totally blocked, just the queues and the CLI. I do have a TDM400 card on one of the machines where it happens, but I have another, bare bone installation with just SIP and IAX2 clients, were I also see it. * 1.2.4. Håkan Källberg pgppTCzzmzDO3.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Trunking, No connections any more...
Hello! I have bin trying to set up trunking between some of my Asterisk boxes but had no luck... I use Asterisk 2.0.6 on SuSE 6.2, but I have had the same problem with erlier releases. I have a working connection and can place multiple calls in both direktions. Than I set trunk=yes on both sides and reload. CLI iax2 show peers shows a (T) and low latency on both sides. Now it is not possible to get a call trough any more:-( I have working timers on both sides, Digium cards or ztDummy. I don't find very much diskussion about problems with trunking accept with the timing. Maybe I have missed something important. Does anyone have an idea??? By the way, I have removed SuSEs precompiled 1.0.0 zaptel drivers and use my own. Thanks in advance! Håkan pgp7mGEOlGQGj.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 trunking - timing - ztdummy??
On Tue, Nov 16, 2004 at 01:56:07PM +0100, Patrick wrote: I don't think you need to do anything with zaptel.conf when you are using ztdummy. Someone on irc mentioned that ztdummy's timing is off by about 3%. So if you are serious about your timing why don't you get a X100P to make sure you have the proper timing. Well, if it works... Compared to the crt package?? [snip] Btw, I am using Asterisk 1.0.2, Zaptel 1.0.0, SuSE 9.2 ( kernel 2.6.8 ) [snip] Afaik it is preferred to always install zaptel, libpri, asterisk and asterisk-addons from the same release. You are using asterisk 1.0.2 with zaptel 1.0.0. Well, what I found as tar.gzed releases Get the latest stable branch with: cvs co -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds Install them and see if that solves anything. I will try! As I read the download page on asterisk.org I now discovered the following: Installation should be in the order: zaptel, libpri, Asterisk This might be my problem! As I build my own packages I like to see things as independent installations, as long as they come on the right places. I will try to build them all, make a fake installation ( in the right order ) and build a package out of that. Or do you thing that I need to have the zaptel installed before I even build Asterisk?? Thanks very much for your thoughts and help!! Regards:Håkan pgpXm01IMuCCy.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 trunking - timing - ztdummy??
On Tue, Nov 16, 2004 at 03:22:04PM +0100, Patrick wrote: On Tue, 2004-11-16 at 14:39 +0100, Håkan Källberg wrote: [snip] Well, if it works... Compared to the crt package?? Yes it does work. I use ztdummy myself on my hardly loaded home * box because I am too lazy to plug my X100P back in :) The 3% off will have a negative impact when you start to get a serious amount of meetme's or try to iax2-trunk a lot of calls to another * box. So at low load it will not be noticeable but at higher load it will have an impact (so I was told). What do you mean with crt package? Ohh, sorry, rtc ( real time clock ) I mean... Do I need the X100P in both ends in the case of trunking? Only one side complains. Yes, the build (make install) order is: 1) zaptel 2) libpri 3) asterisk 4) asterisk-addons 5) asterisk-sounds Yes, I see now, the Makefile in asterisk looks for an zaptel installation. Regards:Håkan pgpD7erf3ry3n.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 trunking - timing - ztdummy??
Hello! Perhaps someone can spread i little bit light on this: I want to trunk two Asterisk systems with each other. System A, behind a NAT-Firewall and System B with a real IP address. aix.conf on B: [mytrunk] host=dynamic username=mytrunk auth=md5 secret=yyy trunk=yes iax.conf on A: register = mytrunk:[EMAIL PROTECTED] When I make a reload an B I get the following: Nov 15 16:32:32 WARNING[-1244329040]: chan_iax2.c:6427 build_peer: Unable to support trunking on peer 'mytrunk' without zaptel timing I have downloaded the zaptel package, compiled it ( including ztdummy, which may be what I need ) and installed it. The kernel modules load: ztdummy 3492 0 zaptel228996 1 ztdummy crc_ccitt 2176 1 zaptel I don't know how to configure zaptel ( /etc/zaptel.conf ) to get this to work. I have no hardware, I only want timing for the IAX2 trunk ( and later on for Conference calls ). I have also read about the rtc package but have not tried it. I may have overseen very basic things... Please enlighten me! Regards:Håkan pgpQHZNKq9tFm.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 trunking - timing - ztdummy??
On Mon, Nov 15, 2004 at 10:02:43AM -0800, Bob Knight wrote: Håkan Källberg wrote: I want to trunk two Asterisk systems with each other. System A, behind a NAT-Firewall and System B with a real IP address. aix.conf on B: [mytrunk] host=dynamic username=mytrunk auth=md5 secret=yyy trunk=yes iax.conf on A: register = mytrunk:[EMAIL PROTECTED] When I make a reload an B I get the following: Nov 15 16:32:32 WARNING[-1244329040]: chan_iax2.c:6427 build_peer: Unable to support trunking on peer 'mytrunk' without zaptel timing I have downloaded the zaptel package, compiled it ( including ztdummy, which may be what I need ) and installed it. The kernel modules load: ztdummy 3492 0 zaptel228996 1 ztdummy crc_ccitt 2176 1 zaptel I don't know how to configure zaptel ( /etc/zaptel.conf ) to get this to work. I have no hardware, I only want timing for the IAX2 trunk ( and later on for Conference calls ). I have also read about the rtc package but have not tried it. Try running zttest. Once zttest is working you should be OK. # ztest ... 99.975586% 99.975586% --- Results after 246 passes --- Best: 100.00 -- Worst: 99.951172 I supose it is working.(?) Directly after this I do: # asterisk -c WARNING[-1211064192]: chan_iax2.c:6609 build_user: Unable to support trunking on user 'mytrunk' without zaptel timing Wat givs??? Btw, I am using Asterisk 1.0.2, Zaptel 1.0.0, SuSE 9.2 ( kernel 2.6.8 ) A Pentium with lots of GHz and plenty Mbytes RAM, on the server side, lesser hardware but same software on the clientside. I see nothing obvios to configure in /etc/zaptel.conf(?) Regards:Håkan pgpsFf9j0ig2D.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: AW: [Asterisk-Users] Firefly 1.9.6 released
On Mon, Nov 01, 2004 at 10:00:35AM +1100, Tim Robbins wrote: Robert Berg wrote: We have had some problems registering the firefly with the Asterisk 1.0.2 it seams that IAX version doesn't match? How to solve this? Can you provide a little more information on the problems you're having with registration? Error messages, from either or both the Asterisk and Firefly sides of the connection, would be most useful. Ethereal traces would also be good. I was involved in these tests too - sitting at the Asterisk end. I have no knowledge about, or access to Windows computer, so I havn't seen the Firefly in real life. Anyway: It was possible to connect Firefly to the Firefly network ( green lamp ). It was not possible to connect Firefly to the cts-au.freshtel.net network using IAX. It was not possible to connect to our own Asterisk/IAX-server. No traffic at all reached us! But it was perfectly possible to connect to the same Asterisk server using SIP. The third party version of Firefly was used, freshly downloaded. No error ( or other ) messages at either end! As we understand, it *should* be possible to connect to an Asterisk server using IAX2, shouldn't it? It seams not to be anything more to configure than hostname, username and password. How can this go wrong?? Thanks in advance! Håkan pgpNbFPlj5UcG.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
