Re: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?
Hi,thanks Jean-Yves, but I've already found that page (googling), but I asked because following those instruction I couldn't find the SIP settings.Maybe are not present on my N70?Well I'll investigate*## on my mobile says: V 2.0539.1.219-10-05RM-84Any hints?Thanks2006/8/1, Jean-Yves Avenard [EMAIL PROTECTED]: HiOn 8/1/06, FaberK [EMAIL PROTECTED] wrote: Hi folks, I got an N70. Any lynks for the voip/sip configuration? Thanks .:FaberK:.they aren't hard to find !this one works for me:http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html One note of warning :the Nokia will not work if behing NAT ... I've tried everything butI've never managed to get it to work unless the Nokia had a public IPaddress or was on the same subnet as the asterisk server. Be interested to know if you can find a way around thisJY___--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 -- .:FaberK:. ___ --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] Re: If you prefer to read this mail list asa forum ...
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... The only thing I have noticed is that some of my posts do not make it to the list, so I send many of my posts directly to the list. I have the same situation right here. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Re: question about asterisk DB
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Check this for a detailed description: http://en.wikipedia.org/wiki/Berkeley_DB Copy/paste Berkeley DB (DB) is a high-performance, embedded database library with bindings in C, C++, Java, Perl, Python, Tcl and many other programming languages. DB stores arbitrary key/data pairs, and supports multiple data items for a single key. DB can support thousands of simultaneous threads of control manipulating databases as large as 256 terabytes, on a wide variety of systems including most UNIX-like and Windows systems as well as real-time operating systems. Well, it seams I can store 1000 Caller ID records (name + number). Thank you for link. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Re: Re: Re: TE420P/TE415P?
On Mon, Jul 31, 2006 at 05:24:02PM -0400, Matt Florell wrote: On 7/31/06, Julio Arruda [EMAIL PROTECTED] wrote: Matt Florell wrote: Yes, that is very confusing :) Is there no way to throw a timer chip in there(I suppose it's way too late to put that suggestion forward now)? Curiosity, isn't the timer from the 2.6 kernel 'good enough' for Asterisk purposes nowadays ? Or there is a constraint using 2.6+ztdummy that is not obvious (to me at least :-)) ? It can be a very confusing set of steps to get ztdummy installed properly depending on the version of 2.6 kernel that you are using, and it is usually not as accurate of a timer as a zaptel hardware timer is. ztdummy ialso workss with older 2.6 kernels without USE_RTC . ztdummy's USE_RTC mode is is not exactly perfect, as it simply drops a tick 24 out of every 1024 ticks. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Re: question about asterisk DB
On Tue, Aug 01, 2006 at 08:07:01AM +0200, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Check this for a detailed description: http://en.wikipedia.org/wiki/Berkeley_DB Copy/paste Berkeley DB (DB) is a high-performance, embedded database library with bindings in C, C++, Java, Perl, Python, Tcl and many other programming languages. DB stores arbitrary key/data pairs, and supports multiple data items for a single key. DB can support thousands of simultaneous threads of control manipulating databases as large as 256 terabytes, on a wide variety of systems including most UNIX-like and Windows systems as well as real-time operating systems. Well, it seams I can store 1000 Caller ID records (name + number). Thank you for link. http://en.wikipedia.org/wiki/Berkeley_DB#Licensing Copy/paste Versions 2.0 and higher of Berkeley DB are available under a dual license (see http://www.sleepycat.com/download/licensinginfo.shtml). Versions earlier than 2.0 are available under the BSD license, which means free use commercially Asterisk, like glibc, cannot use those later versions and uses 1.x . Check the docs more carefully. Still, 1000-s of records shouldn't be a problem. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] VoipNow 1.2.0 Beta
On 14:44, Mon 31 Jul 06, Tom wrote: Any good suggestions on where to buy rack space in a country that is not honoring stupid US patent law and has great and secure Internet connections? Easyspeedy (denmark) Server4you (germany) Those two are cheap and give you a lot of stuff. Connection is real good. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmd DIAL - Who picked up the call?
I have had a similar problem a few days ago, when i did a blindtransfer i wanted to know which extension the transferer had. i added a variable my self: pbx_builtin_setvar_helper(chan, BLINDTRANSFERER, transferee-cid.cid_num); i see that this is not what YOU need, but maybe it helps to get an idea. btw. this is not directly connected to your problem, but: when you park a call (asterisk feature defautl keys: #700 ...) at your isdn phone and you forgot to catch the call on another phone, the phone from where you parked the call, should ring after 45 seconds (default) does this work for you? (which asterisk version dou you have?) regards KAI ___ --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] SRTP help
Is SRTP available in asterisk? Or how to implement it ? am using trixbox Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Permission for files generated by voicemail
Hi
There is a problem in Asterisk 1.2.10 (at least). Even though in
theorie the source code of app_voicemail.c can be modifier to set up
the proper permission on the directories and file created for the
voicemail, this code can not work.
It doesn't take into account that the umask needs to be set properly
for the argument given to open to act as intended. As a result,
changing the value of VOICEMAIL_FILE_MODE will have no effect in most
cases.
I've adapted a patch that I found earlier which also set-up the group
owner. I've only extracted setting up the permissions as that's all I
needed and starting asterisk with the right group permission does the
job just as well.
Is there a centralized way to post all those patches? I have a few
more in the pipeline ...
Thanks
JY
diff -r -u asterisk-1.2.10/apps/app_voicemail.c asterisk-1.2.10-umask/apps/app_voicemail.c
--- asterisk-1.2.10/apps/app_voicemail.c 2006-07-14 07:22:11.0 +1000
+++ asterisk-1.2.10-umask/apps/app_voicemail.c 2006-08-01 18:24:08.0 +1000
@@ -74,9 +74,12 @@
#include asterisk/res_odbc.h
#endif
+#include pwd.h
+#include grp.h
+
#define COMMAND_TIMEOUT 5000
-#define VOICEMAIL_DIR_MODE 0700
-#define VOICEMAIL_FILE_MODE 0600
+#define VOICEMAIL_DIR_MODE 0770
+#define VOICEMAIL_FILE_MODE 0660
#define VOICEMAIL_CONFIG voicemail.conf
#define ASTERISK_USERNAME asterisk
@@ -421,6 +424,36 @@
LOCAL_USER_DECL;
+static void set_owner_and_group_all(const char* dir, int msgnum)
+{
+ DIR *vmdir = NULL;
+ struct dirent *vment = NULL;
+char fn[32];
+ char pn[1024];
+ snprintf(fn, sizeof(fn), msg%04d, msgnum);
+
+ if (sizeof(dir) + 11 = sizeof(pn)) {
+ ast_log(LOG_WARNING, directory name too long to set owner and group, skipping\n);
+ return;
+ }
+ if ((vmdir = opendir(dir))) {
+ while ((vment = readdir(vmdir))) {
+ if (!strncmp(vment-d_name, fn, 7)) {
+strcpy(pn, dir);
+pn[strlen(dir)] = '/';
+pn[strlen(dir)+1] = 0;
+strcat(pn, vment-d_name);
+if (chmod(pn, VOICEMAIL_FILE_MODE)) {
+ast_log(LOG_WARNING, chmod '%s' failed: %s\n,
+ pn, strerror(errno));
+}
+ }
+ }
+ closedir(vmdir);
+ }
+}
+
+
static void populate_defaults(struct ast_vm_user *vmu)
{
ast_copy_flags(vmu, (globalflags), AST_FLAGS_ALL);
@@ -2635,6 +2668,7 @@
rename(tmptxtfile, txtfile);
ast_unlock_path(dir);
+ set_owner_and_group_all(dir, msgnum);
/* Are there to be more recipients of this message? */
while (tmpptr) {
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[asterisk-users] asterisk gui
Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] asterisk gui
try www.trixbox.orgasterisk source does not come with any GUIOn 8/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.All science is either physics or stamp collecting.-- Ernest Rutherford___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gui
Trixbox is not a GUI, it's a package that includes the OS, Asterisk, a GUI, etc. FreePBX is the GUI included in Trixbox.AlexOn 8/1/06, Rajeev Natarajan [EMAIL PROTECTED] wrote: try www.trixbox.orgasterisk source does not come with any GUI On 8/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.All science is either physics or stamp collecting.-- Ernest Rutherford___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --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] SoftHangup with Polycom_acd_functions release of asterisk
Hi,
I trying to get the softhangup option to work. I'm using the
Polycom_acd_functions branch of asterisk, so not sure if it works with this,
or I'm doing something wrong.
Below is what I have in the dial plan, using 444 and a mobile for testing,
as I would like to use this for emergency services. The pstn-spa3k2 is a
Linksys 3000 ATA.
[emergency]
exten = 444,1,Macro(emergencyoutbound,${EXTEN},60)
[macro-emergencyoutbound]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],60,) currently mobile number
for testing
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Congestion()
exten = s-BUSY,1,Busy()
exten = s-CONGESTION,1,Congestion()
exten = s-CHANUNAVAIL,1,Goto(s,300)
exten = s,300,SoftHangup(SIP/pstn-spa3k2)
exten = s,301,Dial(SIP/[EMAIL PROTECTED],60,)
Does this look right?
Thanks,
Dean.
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Re: [asterisk-users] MWI from Asterisk to Meridian
On Monday 31 July 2006 16:32, kritikus Araklidas wrote: Anyone know some idea if the Asterisk voicemail (WMI) can send the messages to meridian for activate the light on meridian digital phones for voicemail notification Aside from using a Norstar ATA connected to an FXS port on Asterisk and executing a hookflash *1, no. There isn't a really good way to do it, as that is part of what keeps you hooked into their proprietary crap. It's the same as trying to tie in SIP phones through Asterisk to a Norstar system. I have it done through a PRI but even then Norstar sees the phones as external destinations, so I can't pick up Norstar parked calls and can't transfer calls over. -A. ___ --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] Re: Re: FYI - first release of alarm response code.
If you could just post a link to your source after it is done, that would be great. My need would be tied to the voicemail and if I could use that instead of a database (for the most part), I think it would be preferred and more portable. 1. Be triggered by a script that monitors a VM folder every few minutes for new VMs. (Inbox) 2. See if we have already called someone and call the next user. 3. The call would be a connection with the callee and VoicemailMain([EMAIL PROTECTED]) 4. Repeat every few minutes. If someone listens to the message, it will get moved to the Old VM folder and no longer trigger calls. I could use variables in my script for the destination and a text file for who has been called last. My other issue is that the first number called is a pager which is passed between the techs when they are on call. As for the ticket system or monitoring systems. (the above scenario is for a user to call when no IT staff is in the building) nagios: I have seen a reference to someone that used a shell script to have nagios use a local (on monitoring box) copy of festival and record the problem's text as audio, then make a SIP or IAX call and call a tech with the message. OTRS: I could also add a line in my script to send an email to my ticket system. When the tech has dealt with the issue, there is an open ticket to be filled out and closed. (actually, I would prolly just have the VM box email the notification with the VM attached) -- -- Steven http://www.glimasoutheast.org Kevin Withnall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Ill actually be working on this shortly. Ive already designed the system in my head and just need to write it now :-) The problem is it will be fairly integrated into the alarm response code. I don't think it would be hard to write a phpagi script to do this normally. It would just have to... 1. be triggered by a call file in outgoing 2. run the programm see if theres an ack from the user if so, write an ack to the database if not, lookup the next person in the database and write another call file We have a web based job system here that would probably benefit from such a feature. Once the alarm code is written, ill look at this code if you like. Im not the best programmer in the world but I like being able to contribute to the asterisk community. See ya The ability to 'sequentially' call responders instead of calling all at once This is a feature that I would like to see integrated into the voicemail system. We have an extension now, that when someone leaves a message, it calls a different number every 5 minutes, until someone actually listens to the message. This is done in case the on call person fails to get the call, it will go to me next, then to members of my staff. This is working on an old PhoneXpress that I am hoping to phase out and get that feature into asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nat and qualify questions
Are there any problems with always having nat=yes and qualify=yes? We just opened up our server to be accessible to SIP from the internet. (used to require VPN) I had to set the SIP setting for my test softphone to nat=yes and qualify=yes. This makes sense. Some of these phone will never leave our building. Some of these phone will come and go. (laptops) Is the any negatives to just have all phones set to nat=yes and qualify=yes? If not, why is it not the default? Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --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] asterisk gui
true - i was meaning to say that it has a gui 'bundled' with it... (not to mention phpmyadmin, AGI to connect to high-level application development tools such as PHP and Perl, integrated voicemail and fax-to-email support, contact management, calling card billing and management software. autoconfiguration for Digium and Cisco phone hardware, an integrated text-to-speech system) :)mea culparajeevOn 8/1/06, Alex Robar [EMAIL PROTECTED] wrote: Trixbox is not a GUI, it's a package that includes the OS, Asterisk, a GUI, etc. FreePBX is the GUI included in Trixbox.AlexOn 8/1/06, Rajeev Natarajan [EMAIL PROTECTED] wrote: try www.trixbox.orgasterisk source does not come with any GUI On 8/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.All science is either physics or stamp collecting.-- Ernest Rutherford___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gui
Well that made it sound like a much better system than I did ;-)AlexOn 8/1/06, Rajeev Natarajan [EMAIL PROTECTED] wrote:true - i was meaning to say that it has a gui 'bundled' with it... (not to mention phpmyadmin, AGI to connect to high-level application development tools such as PHP and Perl, integrated voicemail and fax-to-email support, contact management, calling card billing and management software. autoconfiguration for Digium and Cisco phone hardware, an integrated text-to-speech system) :)mea culparajeevOn 8/1/06, Alex Robar [EMAIL PROTECTED] wrote: Trixbox is not a GUI, it's a package that includes the OS, Asterisk, a GUI, etc. FreePBX is the GUI included in Trixbox.AlexOn 8/1/06, Rajeev Natarajan [EMAIL PROTECTED] wrote: try www.trixbox.orgasterisk source does not come with any GUI On 8/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.All science is either physics or stamp collecting.-- Ernest Rutherford___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --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] MWI from Asterisk to Meridian
Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --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] Is there a smarter way to ban expensive calls in dial plan?
Hi List, I need a bit of advice please. I want to ban calls to expensive destinations such as cell phones. This is fairly simple here in the UK because all cell phone numbers begin with a 7 where as all geographic numbers begin 1 and 2 Elsewhere this is different, take Andorra for example all numbers begin 376, cell phone numbers are 3763, 3764 and 3765 So if I try the following dial plan my pattern always matches the first wild card Exten = _00376.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion I seem to have been able to fix this with adding an x after the 6 in the first extension to make the patterns all the same length and thus making a better match with the blocked numbers. Example: Exten = _00376x.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion This is just so long winded, and you can imagine doing this for a huge list of destinations. If any one can suggest an improved or more efficient way of doing this, I would be greatly appreciated! Best regards Chris -- Chris Blunt Entropy IT Ltd ___ --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] Is there a smarter way to ban expensive calls in dial plan?
Insert your patterns in a database, have a field called expensive, and query your database before making a call!On 8/1/06, Chris Blunt [EMAIL PROTECTED] wrote: Hi List, I need a bit of advice please. I want to ban calls to expensive destinations such as cell phones. This is fairly simple here in the UK because all cell phone numbers begin with a 7 where as all geographic numbers begin 1 and 2 Elsewhere this is different, take Andorra for example all numbers begin 376, cell phone numbers are 3763, 3764 and 3765 So if I try the following dial plan my pattern always matches the first wild card Exten = _00376.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion I seem to have been able to fix this with adding an x after the 6 in the first extension to make the patterns all the same length and thus making a better match with the blocked numbers. Example: Exten = _00376x.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion This is just so long winded, and you can imagine doing this for a huge list of destinations. If any one can suggest an improved or more efficient way of doing this, I would be greatly appreciated! Best regards Chris -- Chris Blunt Entropy IT Ltd ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --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] AddQueueMember and Local channel
Hi, I have one fairly basic question about AddQueueMember diaplan application, which I'm sure you guys will know to help me with: If I add Local channel to the queue using AddQueueMember (for example: AddQueueMember(MyQueue,Local/[EMAIL PROTECTED]) ), the newly added queue member will have UNKNOWN status and calls will not be delivered to that member. What must be done, so that this member will get status NOT IN USE (if I use the show queue CLI command terminology) and that calls will be delivered to that member? Regards, Alex ___ --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] Is there a smarter way to ban expensive calls indial plan?
Hi, try to list the blocked numbers first! Then you should be able to use wildcards without a problem. :-) That was the solution for the same problem at our dialplan. hth Martin - Original Message - From: Chris Blunt To: asterisk-users@lists.digium.com Sent: Tuesday, August 01, 2006 4:16 PM Subject: [asterisk-users] Is there a smarter way to ban expensive calls indial plan? Hi List, I need a bit of advice please. I want to ban calls to expensive destinations such as cell phones. This is fairly simple here in the UK because all cell phone numbers begin with a 7 where as all geographic numbers begin 1 and 2 Elsewhere this is different, take Andorra for example all numbers begin 376, cell phone numbers are 3763, 3764 and 3765 So if I try the following dial plan my pattern always matches the first wild card Exten = _00376.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion I seem to have been able to fix this with adding an x after the 6 in the first extension to make the patterns all the same length and thus making a better match with the blocked numbers. Example: Exten = _00376x.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion This is just so long winded, and you can imagine doing this for a huge list of destinations. If any one can suggest an improved or more efficient way of doing this, I would be greatly appreciated! Best regards Chris -- Chris Blunt Entropy IT Ltd ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing Fast AGI calling 'h' exten without hanging up
Using the 1.2 branch of SVN, I've been experimenting with FastAGI. I want to do something useful for the caller (e.g. play a message) if the FastAGI server is not running, i.e. AGI gets connect refused. What I have found is that when AGI gets connect refused, it returns -1, and control is passed to the 'h' extension WITHOUT hanging up the channel! My SIP phone thinks it is still connected, but show channels displays no active channels. Putting a Hangup in the 'h' extension doesn't help - the SIP phone still doesn't hang up. I suppose the problem is that AGI would be better returning an AGISTATUS variable instead of just a return valoue of -1. What is the best way to approach this? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] MWI from Asterisk to Meridian
May be you can build an application which controls the background terminal of the Meridian. (This would be a serial connection to the M1) This application sends background commands like: se mw 3000. This could be a try. Best regards Hans Andrew Kohlsmith schrieb: Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help debugging strange asterisk behaviour
Hi, I'm one of those types who want to know what the heck is wrong when something is wrong. I just installed a new server (see config below) and it all works fine for a few hours. But after 3-5 hours asterisk starts behaving VERY strangely for no apparent reason... 1) MoH stops playing 2) Some calls are not hung up from Zap-side 3) Flash Operator Panel starts showing all kind of random letters. 4) Agents are unable to login/logout. ..and so on. But the strange thing is that some things seem to work perfectly fine as usual. Inbound calls are getting playbacks() but no MoH when sent to queue, and caller is not sent to an agent. Outgoing sip and zap calls work fine (until all zapchans are filled because of the above hangup problem which is NOT consistent). I've tried to debug the asterisk log but there are NO ERRORS! I have asterisk installed on a Dell 2850 server with dual Xeon CPU's. I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with freepbx-2.1.1 ontop of it all. I would really appreciate some thoughts on this. Please ask me for furhter info if needed since I'm no debugger. It's a hell of a task to reinstall the whole server so I'd like to know what went wrong this time first. Regards, Jan ___ --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] Missing Fast AGI calling 'h' exten without hanging up
Tony Mountifield wrote: Using the 1.2 branch of SVN, I've been experimenting with FastAGI. I want to do something useful for the caller (e.g. play a message) if the FastAGI server is not running, i.e. AGI gets connect refused. What I have found is that when AGI gets connect refused, it returns -1, and control is passed to the 'h' extension WITHOUT hanging up the channel! My SIP phone thinks it is still connected, but show channels displays no active channels. Putting a Hangup in the 'h' extension doesn't help - the SIP phone still doesn't hang up. I suppose the problem is that AGI would be better returning an AGISTATUS variable instead of just a return valoue of -1. What is the best way to approach this? What sip phone? The linksys spa942 for some reason does not react to bye sip messages. ___ --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] Media direct from IAX Phone to IAX Phone
HI I want to route media directly to one Caller IAX Phone to Called IAX phone signaling IAX Phone1-Asterisk---IAX Phone2 and media IAX Phone1IAX Phone2 Is it possible ? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] SV: Help debugging strange asterisk behaviour
Actually I found one error now after a reboot..Although I don't think it has anything to do with the strange behaviour. Could someone please tell me what this means? Aug 1 16:59:25 DEBUG[6771] chan_zap.c: Failed to read gains: Invalid argument Where is the invalid argument? I've set the gains in zapata.conf to rxgain=-1.0 txgain=-1.5 Regards, Jan -Ursprungligt meddelande- Från: Jan Sarin Skickat: den 1 augusti 2006 17:12 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Help debugging strange asterisk behaviour Hi, I'm one of those types who want to know what the heck is wrong when something is wrong. I just installed a new server (see config below) and it all works fine for a few hours. But after 3-5 hours asterisk starts behaving VERY strangely for no apparent reason... 1) MoH stops playing 2) Some calls are not hung up from Zap-side 3) Flash Operator Panel starts showing all kind of random letters. 4) Agents are unable to login/logout. ..and so on. But the strange thing is that some things seem to work perfectly fine as usual. Inbound calls are getting playbacks() but no MoH when sent to queue, and caller is not sent to an agent. Outgoing sip and zap calls work fine (until all zapchans are filled because of the above hangup problem which is NOT consistent). I've tried to debug the asterisk log but there are NO ERRORS! I have asterisk installed on a Dell 2850 server with dual Xeon CPU's. I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with freepbx-2.1.1 ontop of it all. I would really appreciate some thoughts on this. Please ask me for furhter info if needed since I'm no debugger. It's a hell of a task to reinstall the whole server so I'd like to know what went wrong this time first. Regards, Jan ___ --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] nat and qualify questions
As far as i know qualify=yes will increase you network traffic, this will make asterisk to communicate with all sip friends every X seconds, not sure the default value.On 8/1/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: Are there any problems with always having nat=yes and qualify=yes? We just opened up our server to be accessible to SIP from the internet. (used to require VPN) I had to set the SIP setting for my test softphone to nat=yes and qualify=yes. This makes sense. Some of these phone will never leave our building. Some of these phone will come and go. (laptops) Is the any negatives to just have all phones set to nat=yes and qualify=yes? If not, why is it not the default? Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --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] Re: Missing Fast AGI calling 'h' exten without hanging up
In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Using the 1.2 branch of SVN, I've been experimenting with FastAGI.
I want to do something useful for the caller (e.g. play a message)
if the FastAGI server is not running, i.e. AGI gets connect refused.
What I have found is that when AGI gets connect refused, it returns -1,
and control is passed to the 'h' extension WITHOUT hanging up the channel!
My SIP phone thinks it is still connected, but show channels displays
no active channels.
Putting a Hangup in the 'h' extension doesn't help - the SIP phone still
doesn't hang up.
I suppose the problem is that AGI would be better returning an AGISTATUS
variable instead of just a return valoue of -1.
What is the best way to approach this?
What sip phone?
It's a Grandstream BT102. I did a SIP debug, and whereas calling Hangup from
the dialplan generates a SIP BYE, which hangs up the phone normally, if I
call AGI(agi://localhost/foo) and the server isn't running, there is no
SIP BYE, only a SIP CANCEL.
Actually, writing that made me wonder. My dialplan was like this:
exten = 8008,1,Answer
exten = 8008,n,AGI(agi://localhost/foo)
exten = 8008,n,Playback(vm-goodbye)
exten = 8008,n,Hangup
exten = h,1,NoOp(Hangup in context ${CONTEXT})
I've just gone back and added a Wait(0.5) between the Answer and AGI, and
now the phone gets correctly hung up when AGI returns -1. So the SIP issue
was presumably that the call setup hadn't completed when it was hung up,
and nothing to do with AGI. I get exactly the same effect with this:
exten = 8009,1,Answer
exten = 8009,n,Hangup
The remaining issue is probably just a design change in AGI: if it gets
a connection refused, or even some other error, it would be more useful
to set a channel variable instead of just hanging up, so that the dialplan
could take a fallback path with the call still live.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Park / ParkAndAnnounce
Hi, I have a general Park and Announce question I can't seem to find the answer to. I keep seeing example conf files for ParkAndAnnounce but I'm fairly new to asterisk and I am not sure whether Park and Announce is a replacement for Park or a compliment. I guess my question is, how do I use it? should I just add the lines to my entensions_additional.conf or does this replace the stuff in features.conf? I tried googling, old forum archives and looking in the wiki, but all this stuff assumed I knew more than I actually do about Asterisk. Anyone can point me in the right direction or to the right docs? ___ --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] Problem with distortion of initial voicemail prompt
I'm having a problem where the very first words of the Asterisk voicemail system prompt are distorted into a loud ear-splitting beep. When I dial my VoiceMailMain extension I get this loud beep followed by the rest of the initial voicemail system prompt. After that everything works fine. I've have this problem under both v1.2.6 (self-compiled) and now under 1.2.10 (under Astlinux). My handset is connected to my asterisk box through an iaxy. With the exception of this voicemail prompt problem everything else seems to work fine. The relevant portions of my voicemail.conf and extensions.conf are list below: Voicemail.conf [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp [general] format=wav|gsm|wav49 serveremail=astlinux attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 emaildateformat=%A, %B %d, %Y at %r tz=eastern saycid=yes [default] 1000 = 07055,John Q Public,[EMAIL PROTECTED],,tz=eastern [other] Extensions.conf [general] static=yes writeprotect=yes autofallthrough=yes clearglobalvars=no [globals] [default] [1000] type=friend host=dynamic context=context1 secret=password mailbox=1000 dtmfmode=rfc2833 allow=ulaw insecure=very exten = 8500,1,VoiceMailMain,1000 exten = 8500,2,Hangup [iaxy] type=friend host=dynamic context=home secret=iaxy dtmfmode=rfc2833 mailbox=1000 allow=ulaw insecure=very ___ --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] SendText() displaying text messages onaSIPhandset's screen
Actually that worked perfectly, now I have another issue. I don't know the parking system too well. I'm not sure whether I should hack res_features.c to include a ast_sendtext() call to peer to send the message or if I can do it from the conf file through SendText(). the issue is whether the conf file knows who parked the call in order to send them the message or whther this is something that happens in res_features.c On 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message -From: Guillermo Roditi[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List - Non-CommercialDiscussion [mailto: asterisk-users@lists.digium.com]Sent: Fri, 28 Jul 200617:51:26 -0300Subject: Re: [asterisk-users] SendText() displaying textmessages on aSIPhandset's screen for amessage that says test test -- MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.2.13:32827 ;branch=z9hG4bK.39f5be5f;rport;alias To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE Content-Type: text/plain Max-Forwards: 70 User-Agent: sipsak 0.9.6 From: sip:[EMAIL PROTECTED]:32827;tag=1945b6c2 Content-Length: 9 Content-Disposition: desktop test test Ah, it must be the:Content-Disposition: desktopThat does it... interesting. You may be able to hack chan_sip up a bit and add that header in.Joshua ColpDigium___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Asterisk Server to relay call request
Fadjar
I cannot offer documentation as you request.
In answer to creating a central system. This is possible but requires some
level thought and time.
You may be better choosing one of the turnkey packages available, either
OpenSource or Commercial that if well put together would achieve what you
describe by simple point and drop.
If you want a wide selection I suggest you try the Commercial List.
If you wish to contact me direct : steve 'at ' bicomsystems {dot} com
Steve
- Original Message -
From: Fadjar Tandabawana [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 27, 2006 9:35 AM
Subject: [asterisk-users] Multi Asterisk Server to relay call request
Dear Gurus,
I'm newbe in Asterisk and I want to evaluate the system.
I have several location branch office and I want to use VOIP between them.
Is there any documentation about Asterisk that cover several location and
the dial plan?
Is it possible to have one central Asterisk to control all the remote
asterisk?
Regards,
Fadjar T
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Re: [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer
I explained it backwards,
the thing is I need to make a call right when an event happens, for
example when the second link is down, or when I receive a particular call.
In the following sample, I get a call on the first span E1 (g1), and
transfer it to the second span (g0). IF the link is down, I would
like to call support and let them know.
problem is when line 2 has noanswer line 3 never gets executed.
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call support)
exten = _X.,3,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
exten = _X.,4,Playback(help)
this is another one, that can't make work with the same situation, I
can't hangup the call on the E1 slot without ending the call itself,
I've tested hangup and softhangup
exten =7595,1,answer
exten =7595,2,playback(hello)
exten =7595,3,softhangup(${channel}|a)
exten = 7595,4,Dial(Zap/g0/8734438,60,tr)
exten =7595,4,playback(muchasgracias)
exten =7595,5,hangup
All this to try to do it on the same context, (trying to avoid making a
call file ),
maybe it doesn't make any sense does it?
Manrique Feoli escribió:
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to set up a call on another
E1, if the second call succeds, fine but if the second call
doesn't answer (or if the second E1 link happens to be down)I
can't manage to execute another line of my dialplan to try to setup
the call via another route.
I must be missing something basic.
here are my dialplay lines (taken to the simplest expresion)
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call
support)
exten = _X.,3,Playback(help)
exten = _X.,4,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
Line 2 jumps to the h priority, and doesn't execute line 3.
any clue?
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[asterisk-users] Extend analog phone via SIP (OT)
I'm suddenly needing a way to extend an analog phone extension about 15 miles. One end need to be a phone, SIP or analog, don't care, the other end needs to look like an analog phone to connect to a phone jack on the office PBX. In between the 2 ends is the Internet. I've spent some time looking and the only thing I found that claimed to do this is an analog line extended for $600 the pair. Seems like a SIP phone and an IAXY or a Sipura box should allow me to do this but I can't figure it out. Ira ___ --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] IAX over two T1 connections bad quality
On 31 Jul 2006, at 22:11, Jerry Geis wrote: Help please. I have two systems on the net. one in indiana and one in georgia. connected with IAX. local SIP phones in each office (10 each) are cisco and running sip. TDM04B card in each location has 4 local lines. Incoming calls to each location sound fine always. The problem is dialing between offices the call quality is BAD. Both offices are connected to the net with T1 lines. all data. All phones are setup ulaw 64bit. The IAX connection between the boxes is ulaw 64 bit. I tried skype between the two offices and talked for 15 minutes and had no issue. The machine CPU usage is running 92-97% idle most of the time. Running asterisk 1.2.9.1 and zaptel 1.2.6. There are switches in the mix that have voice traffic having priority. How do I determine what is the issue here? Why is the call quality bad and where is it that I can tweek. How have you configured your switches/routers to give voice priority? Do those rules cover IAX (UDP port 4569)? To see what asterisk thinks the problem is you can look at iax2 show netstats (from memory - I'm off net as I write this) It tells you how many dropped/late/jittered packets it has seen at each end. You probably want to enable the new jitterbuffer (at both ends) if you have not yet. Tim Panton www.mexuar.com ___ --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] Controllable hold music
I remember seeing on a website instructions on how to add controls to hold music (volume, change classes etc.) I've been looking in all the usual places, (voip-info, asteriskguru, asteriskdocs etc.) and I can't find this anywhere. Does anyone know where I can find this? ___ --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] Codec selection / IAX tunnels
I use a provider, that allows me to use IAX tunnelling. If I forward a call that uses G.729 and they are configured to allow G.729 and ulaw, then ulaw will be negotiated (and the call is transcoded). If I forward a call that uses G.729 and they are only configured to use G.729, then (as expected) the call is transmitted using G.729. Is there any way I can force the provider to accept G.729 for some calls, and G.711 for others? This appears to work if I have 2 tunnels set up, (one that is g729 only, and one that is ulaw only), but I don't know if there are any undesirable consequences (other than using more bandwidth). Is there a more sensible approach to this? ___ --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] nat and qualify questions
from http://www.voip-info.org/wiki/view/Asterisk+sip+qualify qualify=xxx|no|yes where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds. If you turn on qualify in the configuration of a SIP device in sip.conf, Asterisk will send a SIP OPTIONS command regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. What happens if you use nat=yes is that Asterisk will consider the IP for communicating with the SIP user agent (UA) as the IP from where the SIP invite comes from instead of taking the one included in the SDP message. Hence if you are using phones inside a LAN this 2 addresses will be the same, but if your SIP UA is outside they will not. Having all your phones set with nat=yes and qualify =yes, will not affect the behaviour of your phones if your network is not really full, but will be a bad and dirty way to do it :)Alyed Return-Path: [EMAIL PROTECTED] Tue Aug 01 08:35:05 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Tue, 1 Aug 2006 08:35:05 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id A1691C3F6; As far as i know qualify=yes will increase you network traffic, this will make asterisk to communicate with all sip friends every X seconds, not sure the default value.On 8/1/06, BerkHolz, Steven [EMAIL PROTECTED] wrote:Are there any problems with always having nat=yes and qualify=yes? We just opened up our server to be accessible to SIP from the internet. (used to require VPN) I had to set the SIP setting for my test softphone to nat=yes and qualify=yes.This makes sense. Some of these phone will never leave our building.Some of these phone will come and go. (laptops) Is the any negatives to just have all phones set to nat=yes and qualify=yes?If not, why is it not the default? Thank You,Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.comBoard member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --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] cmd DIAL - Who picked up the call?
On Tuesday, August 01, 2006 9:36 AM Kai Ober wrote: when you park a call (asterisk feature defautl keys: #700 ...) at your isdn phone and you forgot to catch the call on another phone, the phone from where you parked the call, should ring after 45 seconds (default) does this work for you? (which asterisk version dou you have?) 1.2.9.1 bristuffed and no it does not seem to work. It seems to mixup src and dst channel: == Parked Zap/4-1 on 701. Will timeout back to extension [from_internalisdn] s, 1 in 300 seconds The call came from another extension and another context. Therefore the callback will fail (and _does_ fail)... Will you file a bug report and give me the bug number? ___ --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] IAX and Accountcode
Does the accountcode from a SIP user agent get passed to IAX when trunking a call from one asterisk box to another? The SIP caller id, extension etc do get passsed, so why not the account code? It's a standard field. Doing a 'iax2 debug' doesn't even show the accountcode field. Good grief. IAX2 is really lacking in some areas. There's no way to pass variables between asterisk systems (might be something considered as a requirement for 'enterprise grade' and it doesn't look as if the accountcode (which is kinda important) gets passed through the IAX2 protocol either. Doug. ___ --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] MWI from Asterisk to Meridian
Yeah is true.but we have to sincronize this console command with Asterisk SIP MWI Regards. Cris. From: Johann Steinwendtner [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Asterisk to Meridian Date: Tue, 01 Aug 2006 17:06:41 +0200 May be you can build an application which controls the background terminal of the Meridian. (This would be a serial connection to the M1) This application sends background commands like: se mw 3000. This could be a try. Best regards Hans Andrew Kohlsmith schrieb: Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] IAX and Accountcode
Douglas Garstang wrote: Does the accountcode from a SIP user agent get passed to IAX when trunking a call from one asterisk box to another? The SIP caller id, extension etc do get passsed, so why not the account code? It's a standard field. Doing a 'iax2 debug' doesn't even show the accountcode field. Err, does accountcode get passed when terminating with SIP? I thought accountcode was only used for local call records. ___ --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] IAX and Accountcode
- Original Message - From: Douglas Garstang [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Tue, 01 Aug 2006 15:14:51 -0300 Subject: [asterisk-users] IAX and Accountcode Does the accountcode from a SIP user agent get passed to IAX when trunking a call from one asterisk box to another? The SIP caller id, extension etc do get passsed, so why not the account code? It's a standard field. Doing a 'iax2 debug' doesn't even show the accountcode field. caller id and extension are part of a regular phone call, need to know where it came from and where it's going. Those are part of every protocol. Good grief. IAX2 is really lacking in some areas. There's no way to pass variables between asterisk systems (might be something considered as a requirement for 'enterprise grade' and it doesn't look as if the accountcode (which is kinda important) gets passed through the IAX2 protocol either. IAX2 was designed to leverage the core capabilities of Asterisk and not take the same route that other protocols took by learning from their mistakes. There are just some things it wasn't designed to do 'nor does it claim to do. It wasn't made with the capability to transport accountcode or other arbitrary Asterisk specific information. Could it be added though? sure. Doug. Joshua Colp Digium ___ --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] MWI from Asterisk to Meridian
From voicemail.conf: ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp Maybe this approach can send the commands to the M1. Best regards Hans kritikus Araklidas schrieb: Yeah is true.but we have to sincronize this console command with Asterisk SIP MWI Regards. Cris. From: Johann Steinwendtner [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Asterisk to Meridian Date: Tue, 01 Aug 2006 17:06:41 +0200 May be you can build an application which controls the background terminal of the Meridian. (This would be a serial connection to the M1) This application sends background commands like: se mw 3000. This could be a try. Best regards Hans Andrew Kohlsmith schrieb: Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extend analog phone via SIP (OT)
Ira wrote: I'm suddenly needing a way to extend an analog phone extension about 15 miles. One end need to be a phone, SIP or analog, don't care, the other end needs to look like an analog phone to connect to a phone jack on the office PBX. In between the 2 ends is the Internet. I've spent some time looking and the only thing I found that claimed to do this is an analog line extended for $600 the pair. Seems like a SIP phone and an IAXY or a Sipura box should allow me to do this but I can't figure it out. Others have indicated the sipura's can do that, and if my memory serves correctly, specifically the spa3000. If you look around the voxilla.com site, I think you'll find something that describes how to configure the boxes to do that. There is no need for an asterisk box in that config if you don't want it. ___ --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] Re: How to configure NOKIA N70 with Asterisk?
FK == FaberK [EMAIL PROTECTED] writes: FK Hi, thanks Jean-Yves, but I've already found that page (googling), FK but I asked because following those instruction I couldn't find FK the SIP settings. Maybe are not present on my N70? Well I'll FK investigate *## on my mobile says: V 2.0539.1.2 19-10-05 FK RM-84 Any hints? Tools-settings-connections-sip. So far the only problems I've had are the ones which are already well known: No NAT traversal Switching between making calls on WLAN and GSM/UMTS isn't automatic, and it's not just an easy button push either /Benny ___ --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] MySQL 5.0+ and the MySQL addon - Can use stored procedures?
Hello all! I've searched high and low and cannot find any documentation or even examples of the mysql addon to Asterisk being used with stored procedures/functions in MySQL 5.0+ situations. Anyone tried it? I've been able to do a call to a simple procedure that returns only one column in one row, but when I tried to use a stored proc that returns two columns Asterisk doesn't seem to get anything. Any help, suggestions, links, etc would be greatly appreciated. I'll post any progress I make here if anyone's interested. Rushowr ___ --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] Line drops
Hey all experiencing a quirky problem: 1) call comes in on line 1 welcome too foobar 2) another call comes in on another line (line 2) 3) make transfer on line 1... while line 2 rings 3) line 2 drops after line 1 connects via transfer -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil infiltrated . net http://www.infiltrated.net How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --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] Re: Operator Console(s)/Shared Call Appearances
Yes this is what I want. I guess the question is what is the best way to do it? Use a Queue? or something else? On 25 Jul 2006 13:25:45 +0200, Benny Amorsen [EMAIL PROTECTED] wrote: J == Jones [EMAIL PROTECTED] writes: J Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk J and I'm trying to determine the best way to allow our receptionist J to answer certain executives telephone lines. J It seems there are probably two routes, but I'm not sure of the J limitations of each. You could make both the executive and the receptionist phones ring, perhaps with a very low ring tone for the executives. Then the receptionist will take the call whenever possible. If the call needs to go through to the executive, the receptionist can do a direct call just by pressing a button, and a different (perhaps louder) ring tone can play. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX and Accountcode
-Original Message- From: Joshua Colp [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 01, 2006 8:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX and Accountcode - Original Message - From: Douglas Garstang [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Tue, 01 Aug 2006 15:14:51 -0300 Subject: [asterisk-users] IAX and Accountcode Does the accountcode from a SIP user agent get passed to IAX when trunking a call from one asterisk box to another? The SIP caller id, extension etc do get passsed, so why not the account code? It's a standard field. Doing a 'iax2 debug' doesn't even show the accountcode field. caller id and extension are part of a regular phone call, need to know where it came from and where it's going. Those are part of every protocol. Good grief. IAX2 is really lacking in some areas. There's no way to pass variables between asterisk systems (might be something considered as a requirement for 'enterprise grade' and it doesn't look as if the accountcode (which is kinda important) gets passed through the IAX2 protocol either. IAX2 was designed to leverage the core capabilities of Asterisk and not take the same route that other protocols took by learning from their mistakes. There are just some things it wasn't designed to do 'nor does it claim to do. It wasn't made with the capability to transport accountcode or other arbitrary Asterisk specific information. Could it be added though? sure. What about this scenario? User A calls User B. User A and User B are registered on the same Asterisk system. User B does an attended transfer, and transfers the call to user C, who is registered on a different asterisk system. You set the accountcode to be user B's account code, as user B will be responsible and billed for this call leg. You then do a DUNDi lookup, and get an IAX path to user C on the second asterisk system. You dial user C. At this point, no account code was passed with IAX between the two Asterisk systems. How can user B be billed for the call??? Doug. ___ --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] AddQueueMember and Local channel
Or let me rephrase my question: Why is Local/[EMAIL PROTECTED] of status Unknown as you can see from this CLI snapshot (that includes add queue member CLI instruction as well)? What do I have to do to make it available to the callers that call in the queue testQ: asterisk*CLI add queue member Local/[EMAIL PROTECTED] to testQ Added interface 'Local/[EMAIL PROTECTED]' to queue 'testQ' asterisk*CLI show queue testQ testQhas 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (dynamic) (Unknown) has taken no calls yet No Callers Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Tuesday, August 01, 2006 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AddQueueMember and Local channel Hi, I have one fairly basic question about AddQueueMember diaplan application, which I'm sure you guys will know to help me with: If I add Local channel to the queue using AddQueueMember (for example: AddQueueMember(MyQueue,Local/[EMAIL PROTECTED]) ), the newly added queue member will have UNKNOWN status and calls will not be delivered to that member. What must be done, so that this member will get status NOT IN USE (if I use the show queue CLI command terminology) and that calls will be delivered to that member? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN incoming call - inband info and announcements BEFORE ANSWER
Hi all, Is a way to force Asterisk to send DSS1 PROGRESS message to PSTN with indicator: Inband information now available, before call is established (even before ALERTING phase)? I also think that this indicator can be contained in CALL PROCEEDING message. My idea is to play not billed welcome message on Asterisk system. Just now there is incoming SETUP, Asterisk replies with CALL PROCEEDING (without indicator I presume - but I can think only from Asterisk trace, no ISDN tester available at the moment). In ideal case there should be send PROGRESS or CALL PROCEEDING message with that indicator. How to setup this (for PRI and junghanns.net BRI)? === exten = 222000262,1,Playback(welcome,noanswer) ;At this moment would like to send PROGRESS with Inband info now av. exten = 222000262,2,Dial() Used version is 1.2.4-Bristuffed at the moment. Thank you. Michal P.S. I know this is supported only by some telcos, not all, but at the moment would like to cover Asterisk side of the problem. ___ --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] IAX and Accountcode
- Original Message - From: Douglas Garstang [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Tue, 01 Aug 2006 17:08:15 -0300 Subject: RE: [asterisk-users] IAX and Accountcode What about this scenario? User A calls User B. User A and User B are registered on the same Asterisk system. User B does an attended transfer, and transfers the call to user C, who is registered on a different asterisk system. You set the accountcode to be user B's account code, as user B will be responsible and billed for this call leg. You then do a DUNDi lookup, and get an IAX path to user C on the second asterisk system. You dial user C. At this point, no account code was passed with IAX between the two Asterisk systems. How can user B be billed for the call??? I would think that user B would be billed on the originating system, not the system the call ended up at. Doug. Joshua Colp Digium ___ --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] rx_fax problem
hi, rx_fax fails to get fax on a bit noisy lines but real fax devices can do that on the same line with no problem! what's the problem? thanks ___ --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] Is there a smarter way to ban expensive calls in dial plan?
Ciao Chris, So if I try the following dial plan my pattern always matches the first wild card Exten = _00376.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion This is a common pitfall in Asterisk dialplans: Asterisk doesn't try to match your extensions in the order you insert them into your dialplan, but it sorts them out according to its own internal order. See the CLI command show dialplan example to discover how it sorts them. So, how to solve this misunderstanding? You must create other contexts, and include them in your main context. Asterisk will try to match current context's extensions first, and then extensions included from other contexts, in the order you included them. Please refer to http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting for further information. HTH, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Dundi and Dial Arguments
Dundi question:
Is there a way to pass dial arguments to switch = DUNDi as if you were
dialing using Dial(${DUNDILOOKUP(${EXTEN})},,tTwW)?
We were going to impliment DUNDi, but realized we lost the ability to
use the Dial features.
I could just use the DUNDILOOKUP function, but that keeps you from being
able to use alternate routes if DUNDi returns multiple routes.
I've looked through the source code in pbx_dundi.c (cursory glance) but
can't really find where the dial takes place.
Mitch Sharp
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Re: [asterisk-users] IAX and Accountcode
On 1 Aug 2006, at 21:08, Douglas Garstang wrote: What about this scenario? User A calls User B. User A and User B are registered on the same Asterisk system. User B does an attended transfer, and transfers the call to user C, who is registered on a different asterisk system. You set the accountcode to be user B's account code, as user B will be responsible and billed for this call leg. You then do a DUNDi lookup, and get an IAX path to user C on the second asterisk system. You dial user C. At this point, no account code was passed with IAX between the two Asterisk systems. How can user B be billed for the call??? Consider a more common case: User A uses their local Asterisk (B) to call PSTN Number Z via a trunk to a provider gateway running asterisk (C). If Account code is settable remotely, we can't trust it for billing. User A inserts an account code into their IAX message, it travels through both IAX connections and messes up the billing in C ? That's why the account code isn't passed. Tim Panton www.mexuar.com ___ --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] IAX and Accountcode
-Original Message- From: Joshua Colp [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 01, 2006 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] IAX and Accountcode - Original Message - From: Douglas Garstang [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Tue, 01 Aug 2006 17:08:15 -0300 Subject: RE: [asterisk-users] IAX and Accountcode What about this scenario? User A calls User B. User A and User B are registered on the same Asterisk system. User B does an attended transfer, and transfers the call to user C, who is registered on a different asterisk system. You set the accountcode to be user B's account code, as user B will be responsible and billed for this call leg. You then do a DUNDi lookup, and get an IAX path to user C on the second asterisk system. You dial user C. At this point, no account code was passed with IAX between the two Asterisk systems. How can user B be billed for the call??? I would think that user B would be billed on the originating system, not the system the call ended up at. Who gets billed for the call path from user B on system A to user C on system B? ___ --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] Help debugging strange asterisk behaviour
Are you using mpg123 for MoH or native? What's in your musiconhold.conf? [EMAIL PROTECTED] wrote: Hi, I'm one of those types who want to know what the heck is wrong when something is wrong. I just installed a new server (see config below) and it all works fine for a few hours. But after 3-5 hours asterisk starts behaving VERY strangely for no apparent reason... 1) MoH stops playing 2) Some calls are not hung up from Zap-side 3) Flash Operator Panel starts showing all kind of random letters. 4) Agents are unable to login/logout. ..and so on. But the strange thing is that some things seem to work perfectly fine as usual. Inbound calls are getting playbacks() but no MoH when sent to queue, and caller is not sent to an agent. Outgoing sip and zap calls work fine (until all zapchans are filled because of the above hangup problem which is NOT consistent). I've tried to debug the asterisk log but there are NO ERRORS! I have asterisk installed on a Dell 2850 server with dual Xeon CPU's. I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with freepbx-2.1.1 ontop of it all. I would really appreciate some thoughts on this. Please ask me for furhter info if needed since I'm no debugger. It's a hell of a task to reinstall the whole server so I'd like to know what went wrong this time first. Regards, Jan ___ --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 !DSPAM:500,44cf6f0c41131882367086! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] Re: How to configure NOKIA N70 with Asterisk?
Hi,the problem is that I have not the sip choice into my N70 menu.Today I've made an update of the system, now I have:V 5..0609.2.0.1but still no sip.I think is because my mobile has been customized by my telephone company, H3G. I'll investigate.Thanks01 Aug 2006 20:54:53 +0200, Benny Amorsen [EMAIL PROTECTED]: FK == FaberK[EMAIL PROTECTED] writes:FK Hi, thanks Jean-Yves, but I've already found that page (googling),FK but I asked because following those instruction I couldn't find FK the SIP settings. Maybe are not present on my N70? Well I'llFK investigate *## on my mobile says: V 2.0539.1.2 19-10-05FK RM-84 Any hints?Tools-settings-connections-sip. So far the only problems I've had are the ones which are already wellknown:No NAT traversalSwitching between making calls on WLAN and GSM/UMTS isn't automatic,and it's not just an easy button push either /Benny___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- .:FaberK:. ___ --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] Unicall stack, right versions?
Last night I started compiling all the components of the Unicall stack. So far I've been able to successfully do a testcall. A couple of questions: 1) If you download the snapshot libraries, a funcion that used to be called dtmf_put now has been changed to dtmf_tx_put, however the client code from the other library (I forget which one atm) still uses the old name so I had to fix it. 2) the Makefile patch for the Asterisk channel seems to be for the 1.1.x versions of Asterisk. In the snapshots there's a patch that seems to be for the 1.2.x versions but I haven't tried it yet. Does it work as is or do I have to patch the patch? for Asterisk 1.2.9? In sum, what is the most up-to-date AND stable combination of libraries for the Unicall stack? P.S. 1: A lot of Unicall seems to be hardcoded in the .h and .c files, like the countries and how they behave... I *might* attempt to do something more flexible if I have time *and* brush up my C which I haven't used much in the last 4 years. P.S. 2: A lot of behavior in the Asterisk ecosystem seems to be replicated over and over in the different parts of the code, for example the reading of configuration files, which each programmer does in their own way. How about some generalized configuration code module? Maybe this question is better for the dev list. BarZ ___ --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] Dundi and Dial Arguments
-Original Message-
From: Mitch Sharp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dundi and Dial Arguments
Dundi question:
Is there a way to pass dial arguments to switch = DUNDi as
if you were
dialing using Dial(${DUNDILOOKUP(${EXTEN})},,tTwW)?
We were going to impliment DUNDi, but realized we lost the ability to
use the Dial features.
I could just use the DUNDILOOKUP function, but that keeps you
from being
able to use alternate routes if DUNDi returns multiple routes.
Yes, damn annoying that.
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Re: [asterisk-users] Dundi and Dial Arguments
On 15:39, Tue 01 Aug 06, Douglas Garstang wrote:
-Original Message-
From: Mitch Sharp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dundi and Dial Arguments
Dundi question:
Is there a way to pass dial arguments to switch = DUNDi as
if you were
dialing using Dial(${DUNDILOOKUP(${EXTEN})},,tTwW)?
We were going to impliment DUNDi, but realized we lost the ability to
use the Dial features.
I could just use the DUNDILOOKUP function, but that keeps you
from being
able to use alternate routes if DUNDi returns multiple routes.
Yes, damn annoying that.
I suggest you use an AGI for it.
That gives you way more options
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[asterisk-users] codec conversion
Hello, What is the best utility to convert GSM files into G729 files for batch processing. Thanks WAzb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dundi and Dial Arguments
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dundi and Dial Arguments
On 15:39, Tue 01 Aug 06, Douglas Garstang wrote:
-Original Message-
From: Mitch Sharp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 01, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dundi and Dial Arguments
Dundi question:
Is there a way to pass dial arguments to switch = DUNDi as
if you were
dialing using Dial(${DUNDILOOKUP(${EXTEN})},,tTwW)?
We were going to impliment DUNDi, but realized we lost
the ability to
use the Dial features.
I could just use the DUNDILOOKUP function, but that keeps you
from being
able to use alternate routes if DUNDi returns multiple routes.
Yes, damn annoying that.
I suggest you use an AGI for it.
That gives you way more options
How does AGI help? Your still calling DUNDILOOKUP inside the AGI script, and
not matter how many times you call it, your still always going to get the
lowest priority path returned.
Doug.
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[asterisk-users] Polycom IP600 HTTP Provisioning problem
Hello, The latest Polycom firmware (1.6.x series) supports HTTP(s) provisioning that I have been trying to setup. The admin guide mentions that in the boot settings for the configuration server, URLs of this format can be used - http://user:[EMAIL PROTECTED]/dir/config.cfg But when I use that, the phone seems to be ignoring the sub-dir text and just tries to send requests like these - 172.16.19.160 - - [01/Aug/2006:18:24:54 -0400] GET /bootrom.ld HTTP/1.1 404 288 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) 172.16.19.160 - - [01/Aug/2006:18:24:54 -0400] GET /0004f3445566.cfg HTTP/1.1 404 294 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) 172.16.19.160 - - [01/Aug/2006:18:24:54 -0400] GET /.cfg HTTP/1.1 404 294 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) 172.16.19.160 - - [01/Aug/2006:18:25:07 -0400] GET /0004f2445566-phone.cfg HTTP/1.1 404 300 - Polycom-FileManager/1.0 (libcurl/7.12.1 OpenSSL/0.9.7d) (SIP-1.6.3:0067;SPIPPolycomSoundPointIP-SPIP_601) Which obviously generates a 404. Has anyone tried this with success? The only solution to this that I can think of is to configure a virtual host on the apache side and use a different URL. It would be more convenient if I don't have to create another virtual host on the machine just for the phone configs. Any clues? Thanks, -- VaibhaV Sharma Ishi Systems Inc. http://ishisystems.com ___ --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] Strange behaviour Panasonic KX-TD1232
Ok Ok, the figure doesnt help.Here we go again - -- --- --| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | | Ext1 Ext2Here is my dialplan[incoming]exten = s,1,Answerexten = s,2,Background(prueba-pbx)exten = s,3,Set(TIMEOUT(response)=5)exten = 1001,1,Dial,SIP/1001|20exten = 1001,2,Hangupexten = 1001,102,Congestion,3exten = 1002,1,Dial,SIP/1002|20exten = 1002,2,Hangupexten = 1002,102,Congestion,3[sip]include = outgoingexten = 1001,1,Dial(SIP/1001,20)exten = 1001,2,Hangupexten = 1001,102,Congestion,3exten = 1002,1,Dial(SIP/1002,20)exten = 1002,2,Hangupexten = 1002,102,Congestion,3[outgoing]exten = 0,1,Dial,Zap/g1exten = 0,2,Congestionexten = 0,102,Congestionexten = 9,1,Dial,Zap/g1/9exten = 9,2,Congestionexten = 9,102,CongestionWhen I make a call from PSTN to SIP, first Answer the Panasonic, after this I digit an Extension and the call goes to asterisk, then I dial to sip and the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the call goes to asterisk, then I dial to sip and the call goes on.When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap sending 9 to get PSTN line, the dial the PSTN number and the call goes on.When I make a call from SIP to Ext1 (Ext2 ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything.Your help will be appreciated. ___ --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] Strange behaviour Panasonic KX-TD1232
Again you are not saying how asterisk is connected to the panasonic, stop using pictures. On 8/1/06, Pablo Mora [EMAIL PROTECTED] wrote: Ok Ok, the figure doesn't help. Here we go again… - -- --- -- | SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | | Ext1 Ext2 Here is my dialplan [incoming] exten = s,1,Answer exten = s,2,Background(prueba-pbx) exten = s,3,Set(TIMEOUT(response)=5) exten = 1001,1,Dial,SIP/1001|20 exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial,SIP/1002|20 exten = 1002,2,Hangup exten = 1002,102,Congestion,3 [sip] include = outgoing exten = 1001,1,Dial(SIP/1001,20) exten = 1001,2,Hangup exten = 1001,102,Congestion,3 exten = 1002,1,Dial(SIP/1002,20) exten = 1002,2,Hangup exten = 1002,102,Congestion,3 [outgoing] exten = 0,1,Dial,Zap/g1 exten = 0,2,Congestion exten = 0,102,Congestion exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Congestion exten = 9,102,Congestion When I make a call from PSTN to SIP, first Answer the Panasonic, after this I digit an Extension and the call goes to asterisk, then I dial to sip and the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the call goes to asterisk, then I dial to sip and the call goes on. When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap sending 9 to get PSTN line, the dial the PSTN number and the call goes on. When I make a call from SIP to Ext1 (Ext2… ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything. Your help will be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec conversion
On Tue, 2006-08-01 at 18:23 -0400, Wasif wrote: What is the best utility to convert GSM files into G729 files for batch processing. I don't think sox supports G729. However, you can actually use Asterisk to do this for you if you use the trunk, or upcoming 1.4 release. In the trunk, there is a convert CLI command. First, you will need to download codec_g729a.so from Digium. You will also need some licenses to use it. Then, to convert a directory a bunch of gsm files, you could do something like this ... # for n in `ls *.gsm`; do asterisk -rx convert $n `basename $n .gsm`.g729; done -- Russell Bryant Software Developer Digium, Inc. ___ --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] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
Hi, I could found out why the phone received '404 Not Found'. The reason was this part is not parsed and not Added extensions after that. Because there was not at least one space after ; in front of the line of exten = 0033,1,Meetme(|qM). Regards, Zen From: Zen Kato [EMAIL PROTECTED] Subject: [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found Date: Tue, 01 Aug 2006 12:15:04 +0900 (JST) Hi, I installed asterisk-1.2.10, zaptel-1.2.7 on 2.6.16-1.2108_FC4smp. When I dial '0033', which is a meetme number, but '404 Not Found' comes back. I checked zaptel(ztdummy) on FC4, it seems work fine. Meetme has been working on FC3. Can someone tell me why this happens on FC4? My extensions.conf is; exten = 0033,1,Meetme(|qM) exten = 0033,2,Hangup ngrep shows as follows; U 192.168.0.103:5060 - 192.168.0.3:5070 INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bKa854c86267e80f96..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3 fa865dcc1..To: sip:[EMAIL PROTECTED]:5070..Contact: sip:[EMAIL PROTECTED] 3..Supported: replaces..Call-ID: [EMAIL PROTECTED]: 589 86 INVITE..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: ap plication/sdp..Content-Length: 354v=0..o=0303 8000 8000 IN IP4 192.168. 0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m=audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=r tpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap:2 G726-32/8000..a=rtpmap :15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=20..a=rtpmap:9 G722/16 000..a=ptime:20.. # U 192.168.0.3:5070 - 192.168.0.103:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.103;b ranch=z9hG4bKa854c86267e80f96;received=192.168.0.103..From: sip:[EMAIL PROTECTED] 68.0.3:5070;tag=c7a5ee3fa865dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01 593a47..Call-ID: [EMAIL PROTECTED]: 58986 INVITE..User-A gent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCR IBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]:5070..Proxy-Authenticate: Dige st algorithm=MD5, realm=asterisk, nonce=72494d6d..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bKa854c86267e80f96..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3fa8 65dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Contact: sip:0303@ 192.168.0.103..Call-ID: [EMAIL PROTECTED]: 58986 ACK..U ser-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,C ANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bK6e9ddb4b834276ef..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3 fa865dcc1..To: sip:[EMAIL PROTECTED]:5070..Contact: sip:[EMAIL PROTECTED] 3..Supported: replaces..Proxy-Authorization: Digest username=0303, realm =asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=72494d6 d, response=35378e1d15e71946d8ca187b102d0087..Call-ID: 1c59a92f2174f5ca@ 192.168.0.103..CSeq: 58987 INVITE..User-Agent: Grandstream BT100 1.0.6.8..M ax-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUB SCRIBE..Content-Type: application/sdp..Content-Length: 354v=0..o=0303 8 000 8001 IN IP4 192.168.0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m =audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap: 2 G726-32/8000..a=rtpmap:15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mod e=20..a=rtpmap:9 G722/16000..a=ptime:20.. # U 192.168.0.3:5070 - 192.168.0.103:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.0.103;branch=z9hG4bK6e9ddb4 b834276ef;received=192.168.0.103..From: sip:[EMAIL PROTECTED]:5070;tag=c7a 5ee3fa865dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Call-ID: 1c5 [EMAIL PROTECTED]: 58987 INVITE..User-Agent: Asterisk PBX.. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact : sip:[EMAIL PROTECTED]:5070..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bK6e9ddb4b834276ef..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3fa8 65dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Contact: sip:0303@ 192.168.0.103..Proxy-Authorization: Digest username=0303, realm=asteris k, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=72494d6d, respo nse=9bea041787bf296bcd1c5d730733f615..Call-ID: [EMAIL PROTECTED] .103..CSeq: 58987 ACK..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow:
[asterisk-users] Asterisk with VoIP phone
Hello, Is is possible to setup an asterisk server with out buying Digium card. I mean can we do this type of setup. We all know that X-Lite can be used as a soft phone to have an IP extension. Is it possible to take a service from another VoIP service provider, and get the IP phone number. Make that phone numbe gateway to outside world. Now all the internal extensions use that phone to receive and make calls to out side world. Has any one done this kind of setup or know anything about this. Thank you, -Jai ___ --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] ISDN incoming call - inband info and announcements BEFORE ANSWER
On Aug 1, 2006, at 3:13 PM, Michal Doležel wrote: Is a way to force Asterisk to send DSS1 PROGRESS message to PSTN with indicator: Inband information now available, before call is established (even before ALERTING phase)? I also think that this indicator can be contained in CALL PROCEEDING message. My idea is to play not billed welcome message on Asterisk system. Just now there is incoming SETUP, Asterisk replies with CALL PROCEEDING (without indicator I presume - but I can think only from Asterisk trace, no ISDN tester available at the moment). In ideal case there should be send PROGRESS or CALL PROCEEDING message with that indicator. How to setup this (for PRI and junghanns.net BRI)? Use the Progress() application in your dialplan before you Answer() the line. Use the Background() application with the 'n' flag to play your announcement (so the line is not Answer()'d automatically for you). Matthew Fredrickson ___ --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] VOIP phone for Receptionist use
I've searched through the newsgroup and online and haven't found an answer for my question... maybe I am looking for the wrong terms, I am not sure... I have a client that would like a phone that is like a "typical" receptionists phone. Requirements: - Ability for their3 lines to "light-up" a button on the phone when one of them rings in. - Ability for the phone to ring when the receptionist is on one call and a second or third call is incoming. (this has been the biggest frustration up to now. When a second call comes, there is no tone that heard on the IP500. Perhaps I am missing a setting?) We are currently using: Asterisk @ Home 2.1 Polycom IP500/501 phones Is there a way to do what we need to using the IP500 phones? If so, can anyone give me instructions on how to make it work with [EMAIL PROTECTED]? Thanks for your help in advance. Jeff ___ --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] VOIP phone for Receptionist use
Title: RE: [asterisk-users] VOIP phone for Receptionist use Doesn't [EMAIL PROTECTED] need the DB flag for call waiting disabled? I believe it is *70 to enable call waiting and *71 to disable. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Jeff Busch Sent: Tue 8/1/2006 8:20 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VOIP phone for Receptionist use I've searched through the newsgroup and online and haven't found an answer for my question... maybe I am looking for the wrong terms, I am not sure... I have a client that would like a phone that is like a typical receptionists phone. Requirements: - Ability for their 3 lines to light-up a button on the phone when one of them rings in. - Ability for the phone to ring when the receptionist is on one call and a second or third call is incoming. (this has been the biggest frustration up to now. When a second call comes, there is no tone that heard on the IP500. Perhaps I am missing a setting?) We are currently using: Asterisk @ Home 2.1 Polycom IP500/501 phones Is there a way to do what we need to using the IP500 phones? If so, can anyone give me instructions on how to make it work with [EMAIL PROTECTED] Thanks for your help in advance. Jeff ___ --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] RE: asterisk-users Digest, Vol 25, Issue 2
(Andrew Kohlsmith) wrote: Re: MWI from Asterisk to Meridian So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. Johann Steinwendtner wrote: ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp Can your client accept that, Messages alert from 1 extension, and dial different number to access voicemail? Means omit the VM Call back. Will, like some brand alert from some extensions and vm call back will be different extensions. Nortel side, configured those vm alert port not accept the call from any extension. ( to avoid voicemail call back ) Create speed dial number, let user to access vm from Asterisk. You may look at this link it might help :) http://www.voip-info.org/wiki/index.php?page=Asterisk-Panasonic1232vm ___ --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] ANNOUNCE: libss7
Hey all! For the past year I have been working on and off on an SS7 implementation here at Digium called libss7. I have it to the point where it can pass phone calls, so I figured it would be a good time to release it and let people begin testing it. It's still somewhat bare bones in functionality, but I've been doing a lot of fleshing out of the implementation. Currently, it has been used (making and receiving phone calls) and developed in an ITU SS7 environment, but I have a good chunk of the code included which is required for ANSI support as well. I think I'm going to get an ANSI link in a few weeks, so hopefully I'll have that tested and working relatively soon. It supports MTP2, MTP3, and ISUP. After I get these layers fleshed out, I'm planning on starting on SCCP and the layers above that with the eventual goal of database-lookup and SMS support. To test, you must have a T1/E1 card as well as an SS7 link. You also need to have zaptel installed on your system. Here are the instructions for checking it out of subversion and getting it working: `svn co http://svn.digium.com/svn/libss7/trunk libss7` `cd libss7` `make install` Right now, the changes to chan_zap are implemented in a special developer branch of asterisk. These are the instructions to check it out `svn co http://svn.digium.com/svn/asterisk/team/mattf/asterisk-ss7 asterisk-ss7` `cd asterisk-ss7` If you haven't compiled trunk yet, you may have to run `make` a few times so that the configure script runs and sets things up properly. It should find libss7, and compile chan_zap with support for it. The link is brought up automatically when Asterisk starts. Configuration in zaptel.conf is similar to that of a PRI. Your signalling channel will be set as a dchan and the bearer channels are set as bchan. For information about setting up zapata.conf, see the sample zapata.conf in the configs/zapata.conf.sample in the asterisk-ss7 branch. There also is a libss7 project section on Mantis now for any bugs that you might encounter. Matthew Fredrickson ___ --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] Unicall stack, right versions?
On 8/1/06, Barzilai [EMAIL PROTECTED] wrote: Last night I started compiling all the components of the Unicall stack. So far I've been able to successfully do a testcall. Congratulations! :) 1) If you download the snapshot libraries, a funcion that used to be called dtmf_put now has been changed to dtmf_tx_put, however the client code from the other library (I forget which one atm) still uses the old name so I had to fix it. This does not seems to be a question. But yes, in fact sometimes steve seems forget to update the libraries to match those changes. I had a hard to find problem with logical incorrect argument passing from a function of spandsp used by unicall. 2) the Makefile patch for the Asterisk channel seems to be for the 1.1.x versions of Asterisk. In the snapshots there's a patch that seems to be for the 1.2.x versions but I haven't tried it yet. Does it work as is or do I have to patch the patch? for Asterisk 1.2.9? In sum, what is the most up-to-date AND stable combination of libraries for the Unicall stack? I think the only way to go is actually trying. I doubt someone has made a list of the right versions. Most of people is so happpy of getting unicall finally working that nobody cares wich version they have :p I would recommend use the more recent versions, and only downgrade if you have problems. P.S. 1: A lot of Unicall seems to be hardcoded in the .h and .c files, like the countries and how they behave... I *might* attempt to do something more flexible if I have time *and* brush up my C which I haven't used much in the last 4 years. That would be great :) P.S. 2: A lot of behavior in the Asterisk ecosystem seems to be replicated over and over in the different parts of the code, for example the reading of configuration files, which each programmer does in their own way. How about some generalized configuration code module? Maybe this question is better for the dev list. hum, as far as i know every programmer should be using ast_config() and friends to read configuration files, since the user could choose to use database configuration files, or some other config engine. What do you mean with this? Regards. Moises Silva -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Strange behaviour Panasonic KX-TD1232
Pablo, according to description I assume that you have an FXO at * connected to an FXS port at Panasonic. If this is correct, could you replace Asterisk by a telephone and see if it is possible to make call to Ext1? Jorge Pablo Mora wrote: /Ok Ok, the figure doesn’t help./ / / /Here we go again…/ / / / / / - -- --- --/ /| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN |/ / - -- --- --/ / | |/ /Ext1 Ext2/ / / / / /Here is my dialplan/ / / /[incoming]/ /exten = s,1,Answer/ /exten = s,2,Background(prueba-pbx)/ /exten = s,3,Set(TIMEOUT(response)=5)/ /exten = 1001,1,Dial,SIP/1001|20/ /exten = 1001,2,Hangup/ /exten = 1001,102,Congestion,3/ /exten = 1002,1,Dial,SIP/1002|20/ /exten = 1002,2,Hangup/ /exten = 1002,102,Congestion,3/ / / /[sip]/ /include = outgoing/ /exten = 1001,1,Dial(SIP/1001,20)/ /exten = 1001,2,Hangup/ /exten = 1001,102,Congestion,3/ /exten = 1002,1,Dial(SIP/1002,20)/ /exten = 1002,2,Hangup/ /exten = 1002,102,Congestion,3/ / / /[outgoing]/ /exten = 0,1,Dial,Zap/g1/ /exten = 0,2,Congestion/ /exten = 0,102,Congestion/ / / /exten = 9,1,Dial,Zap/g1/9/ /exten = 9,2,Congestion/ /exten = 9,102,Congestion/ / / /When I make a call from PSTN to SIP, first Answer the Panasonic, after this I digit an Extension and the call goes to asterisk, then I dial to sip and the call goes on successfully. / /When I make a call from Ext1 or Ext2 to SIP, I dial an extension and the call goes to asterisk, then I dial to sip and the call goes on./ /When I make a call from SIP to PSTN, first dial 9 and asterisk gets a Zap sending 9 to get PSTN line, the dial the PSTN number and the call goes on./ /When I make a call from SIP to Ext1 (Ext2… ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything./ / / /Your help will be appreciated./ / / / / / / ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP phone for Receptionist use
By default, CW is turned off in AAH. You need to turn it on. I use the 301, 500, 501, 600, and 601. CW works w/ AAH and Trixbox. You should visit http://www.trixbox.org/index.php if you are using AAH or Trixbox. On 8/1/06, Jeff Busch [EMAIL PROTECTED] wrote: I've searched through the newsgroup and online and haven't found an answer for my question... maybe I am looking for the wrong terms, I am not sure... I have a client that would like a phone that is like a typical receptionists phone. Requirements: - Ability for their 3 lines to light-up a button on the phone when one of them rings in. - Ability for the phone to ring when the receptionist is on one call and a second or third call is incoming. (this has been the biggest frustration up to now. When a second call comes, there is no tone that heard on the IP500. Perhaps I am missing a setting?) We are currently using: Asterisk @ Home 2.1 Polycom IP500/501 phones Is there a way to do what we need to using the IP500 phones? If so, can anyone give me instructions on how to make it work with [EMAIL PROTECTED] Thanks for your help in advance. Jeff ___ --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] Unicall stack, right versions?
Barzilai wrote: Last night I started compiling all the components of the Unicall stack. So far I've been able to successfully do a testcall. A couple of questions: 1) If you download the snapshot libraries, a funcion that used to be called dtmf_put now has been changed to dtmf_tx_put, however the client code from the other library (I forget which one atm) still uses the old name so I had to fix it. Don't use the snapshots. If you use the latest releases this won't happen. 2) the Makefile patch for the Asterisk channel seems to be for the 1.1.x versions of Asterisk. In the snapshots there's a patch that seems to be for the 1.2.x versions but I haven't tried it yet. Does it work as is or do I have to patch the patch? for Asterisk 1.2.9? There hasn't been a need to update the software for some time. The 1.1.x directory works fine with 1.2.x. I should have changed that. Sorry. In sum, what is the most up-to-date AND stable combination of libraries for the Unicall stack? The latest release is, well, the latest release. P.S. 1: A lot of Unicall seems to be hardcoded in the .h and .c files, like the countries and how they behave... I *might* attempt to do something more flexible if I have time *and* brush up my C which I haven't used much in the last 4 years. Bad idea. Its like that for a reason. The present arrangements make support much much simpler. Things like Dialogic, where R2 is alsmost completely configured in config files still end up hard coding a few things. Those config files cause support trouble, though. In my code the variations needed within countries are already allowed for. The whole Unicall scheme is being heavily reworked right now, to separate out the hardware specific elements into their own modules. Hard coded support for countries is something I won't be changing, though. P.S. 2: A lot of behavior in the Asterisk ecosystem seems to be replicated over and over in the different parts of the code, for example the reading of configuration files, which each programmer does in their own way. How about some generalized configuration code module? Maybe this question is better for the dev list. Chaos seems to be the Asterisk way. :-) Steve ___ --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] Strange behaviour Panasonic KX-TD1232
Ok, Im going to stop pictures I have a Digium 4 FXO Card in my asterisk, and connect to Panasonic through 2 extensions (configured in a pool) This means when you dial 200 (example) in Panasonic, the call goes to asterisk and it answers. In this sense, the answer is yes replacing asterisk by a conventional phone, I can dial and the phone rings. The only way in wich call doesnt work is from Sip to Panasonic Ext. I really dont think the problem is asterisk, but ringing cadence and ringback tones from Panasonic. ___ --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] MWI from Asterisk to Meridian
How does the Meridian turn on the MWI? does it use simple DTMF? On 7/31/06, kritikus Araklidas [EMAIL PROTECTED] wrote: Hi everyone: Anyone know some idea if the Asterisk voicemail (WMI) can send the messages to meridian for activate the light on meridian digital phones for voicemail notification Thank Cris. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A2Billing - destination
Caros, I installed the A2Billing - v1.2.2 with Asterisk 1.2.10. All works ok, but when I try callout got a message saying the number in not available. Can you help with a step-by-step to make a card autenticate and dial a number? Thank you Luc Moreira Mais VoIP -- Accepting AUTHENTICATED call from 192.168.0.103: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (gsm), priority = mine -- Executing Answer(IAX2/1003-7, ) in new stack -- Executing Wait(IAX2/1003-7, 2) in new stack -- Executing DeadAGI(IAX2/1003-7, a2billing.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php: line:58 - IDCONFIG : 1 a2billing.php: a2billing.php: line:67 - MODE : standard a2billing.php: a2billing.php: A2Billing AGI internal configuration: a2billing.php: Array a2billing.php: ( a2billing.php: [debug] = 1 a2billing.php: [answer_call] = 1 a2billing.php: [logger_enable] = 1 a2billing.php: [log_file] = /tmp/a2billing.log a2billing.php: [say_goodbye] = a2billing.php: [play_menulanguage] = a2billing.php: [force_language] = EN a2billing.php: [intro_prompt] = a2billing.php: [len_cardnumber] = 10 a2billing.php: [len_aliasnumber] = 15 a2billing.php: [len_voucher] = 15 a2billing.php: [min_credit_2call] = 0 a2billing.php: [min_duration_2bill] = 0 a2billing.php: [notenoughcredit_cardnumber] = 1 a2billing.php: [notenoughcredit_assign_newcardnumber_cid] = 1 a2billing.php: [use_dnid] = a2billing.php: [no_auth_dnid] = Array a2billing.php: ( a2billing.php: [0] = 2400 a2billing.php: [1] = 2300 a2billing.php: ) a2billing.php: a2billing.php: [number_try] = 3 a2billing.php: [say_balance_after_auth] = a2billing.php: [say_balance_after_call] = a2billing.php: [say_rateinitial] = a2billing.php: [say_timetocall] = 1 a2billing.php: [auto_setcallerid] = 1 a2billing.php: [force_callerid] = a2billing.php: [cid_sanitize] = a2billing.php: [cid_enable] = 1 a2billing.php: [cid_askpincode_ifnot_callerid] = 1 a2billing.php: [cid_auto_create_card] = 1 a2billing.php: [cid_auto_assign_card_to_cid] = 1 a2billing.php: [cid_auto_create_card_typepaid] = POSTPAY a2billing.php: [cid_auto_create_card_credit] = 5 a2billing.php: [cid_auto_create_card_credit_limit] = 1000 a2billing.php: [cid_auto_create_card_tariffgroup] = 6 a2billing.php: [callerid_authentication_over_cardnumber] = 1 a2billing.php: [sip_iax_friends] = 1 a2billing.php: [sip_iax_pstn_direct_call_prefix] = 9 a2billing.php: [sip_iax_pstn_direct_call] = 1 a2billing.php: [extracharge_did] = Array a2billing.php: ( a2billing.php: [0] = 091 a2billing.php: ) a2billing.php: a2billing.php: [extracharge_fee] = Array a2billing.php: ( a2billing.php: [0] = 0.25 a2billing.php: [1] = 0.5 a2billing.php: ) a2billing.php: a2billing.php: [dialcommand_param] = |30|HL(%timeout%:61000:3) a2billing.php: [dialcommand_param_sipiax_friend] = |30|HL(360:61000:3) a2billing.php: [switchdialcommand] = 1 a2billing.php: [maxtime_tocall_negatif_free_route] = 5400 a2billing.php: [send_reminder] = a2billing.php: [record_call] = a2billing.php: [monitor_formatfile] = gsm a2billing.php: [base_currency] = usd a2billing.php: [agi_force_currency] = a2billing.php: [currency_association] = Array a2billing.php: ( a2billing.php: [0] = usd:prepaid-dollar a2billing.php: [1] = mxn:pesos a2billing.php: [2] = eur:euro a2billing.php: [3] = all:credit a2billing.php: ) a2billing.php: a2billing.php: [file_conf_enter_destination] = prepaid-enter-dest a2billing.php: [file_conf_enter_menulang] = prepaid-menulang2 a2billing.php: [currency_association_internal] = Array a2billing.php: ( a2billing.php: [usd] = prepaid-dollar a2billing.php: [mxn] = pesos a2billing.php: [eur] = euro a2billing.php: [all] = credit a2billing.php: ) a2billing.php: a2billing.php: ) a2billing.php: a2billing.php: AGI Request: a2billing.php: Array a2billing.php: ( a2billing.php: [agi_request] = a2billing.php a2billing.php: [agi_channel] = IAX2/1003-7 a2billing.php: [agi_language] = en a2billing.php: [agi_type] = IAX2 a2billing.php: [agi_uniqueid] = 1154482698.196 a2billing.php: [agi_callerid] = 1003 a2billing.php: [agi_calleridname] = Luc - Logic Telecom a2billing.php: [agi_callingpres] = 1 a2billing.php: [agi_callingani2] = 0 a2billing.php: [agi_callington] = 0
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
You need to add a ww or 2 like this:
exten = 101,1,Dial(Zap/g1/ww${EXTEN})
or like this:
exten = 9,1,Dial(Zap/g1/ww9)
Hope this helps.
On 8/1/06, Pablo Mora [EMAIL PROTECTED] wrote:
Ok,
I'm going to stop pictures
I have a Digium 4 FXO Card in my asterisk, and connect to Panasonic through
2 extensions (configured in a pool)
This means when you dial 200 (example) in Panasonic, the call goes to
asterisk and it answers.
In this sense, the answer is yes… replacing asterisk by a conventional
phone, I can dial and the phone rings.
The only way in wich call doesn't work is from Sip to Panasonic Ext.
I really don't think the problem is asterisk, but ringing cadence and
ringback tones from Panasonic.
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[asterisk-users] Re: AEL2 Looping
Douglas--
Just to let you know--
- Douglas Garstang dgarstang at oneeighty.com wrote:
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
Here's the output.
The var x never gets incremented! Is this a bug?
The while loops seem to work ok.
I've created bug 7635, then created a branch, found it indeed was a problem
with the spaces in the for() statement,
which is completely stupid, so I put in some code to clear out the spaces, and
then tested in my dialplan.
It worked fine. So, I merged the branch back into trunk, and closed 7635.
So, try it again, after you update your trunk copy, of course, and see
if it's acting more sanely. You should be able to put tabs, spaces,
newlines, and returns in the for, with no problems.
Many thanks for testing out AEL; We are trying to make it solid, but
need help testing from the community. There's always some nits that a
single programmer just plain will not bump into without help.
Now is the time, folks! Play with AEL!!!
murf
--
Steve Murphy
Software Developer
Digium
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Unicall stack, right versions?
Thank you Steve. About the configs in Asterisk... I confess that I'm new to the code so I still need to read more. I didn't know about ast_config() About the hardcodedness of the countries... that seems to be the problem. Everything is too oriented to my country works like this with this telephone company. When in fact, what I'm using is not even to connect it to a the telephone company of my country but to some other machine which has an old Call Center implementation with some other modification of the MF R2 sequence. It doesn't relate specifically to any country. Yes, they are all similar, and being able to specify the number of ANI and DNIS/DID is sometimes all you need, that's why I could make it work. There's some truth in your statement that opening the configuration to external files may get some people into trouble. On the other hand, what I see is a strange mix of: a) If you're doing telephony stuff you should know what you're doing b) Most people using Unicall (Asterisk for that matter) have very little idea of what they are doing and why (copying and pasting configs from here and there). So, where's the sweet spot? :-) I can spend 1 hour reading the source code and finally knowing how to change it to my needs. (For example, adding a new country) Should I need to? Can people from the (b) set do it? Is it scalable? What is more of a support nightmare? Please take all this as constructive comments. I really appreciate your work and if I had to do it from the start it would take me months longer!!! A real question that should go in a different mail, but what the check: Let's say I have two E1 spans, but one needs to talk CountryFooVersion, and the other needs CountryBarVersion (yes, both on the same machine and in the same country, maybe different number of digits for ANI). How would I go about configing that? Thanks BarZ Steve Underwood wrote: Barzilai wrote: Last night I started compiling all the components of the Unicall stack. So far I've been able to successfully do a testcall. A couple of questions: 1) If you download the snapshot libraries, a funcion that used to be called dtmf_put now has been changed to dtmf_tx_put, however the client code from the other library (I forget which one atm) still uses the old name so I had to fix it. Don't use the snapshots. If you use the latest releases this won't happen. 2) the Makefile patch for the Asterisk channel seems to be for the 1.1.x versions of Asterisk. In the snapshots there's a patch that seems to be for the 1.2.x versions but I haven't tried it yet. Does it work as is or do I have to patch the patch? for Asterisk 1.2.9? There hasn't been a need to update the software for some time. The 1.1.x directory works fine with 1.2.x. I should have changed that. Sorry. In sum, what is the most up-to-date AND stable combination of libraries for the Unicall stack? The latest release is, well, the latest release. P.S. 1: A lot of Unicall seems to be hardcoded in the .h and .c files, like the countries and how they behave... I *might* attempt to do something more flexible if I have time *and* brush up my C which I haven't used much in the last 4 years. Bad idea. Its like that for a reason. The present arrangements make support much much simpler. Things like Dialogic, where R2 is alsmost completely configured in config files still end up hard coding a few things. Those config files cause support trouble, though. In my code the variations needed within countries are already allowed for. The whole Unicall scheme is being heavily reworked right now, to separate out the hardware specific elements into their own modules. Hard coded support for countries is something I won't be changing, though. P.S. 2: A lot of behavior in the Asterisk ecosystem seems to be replicated over and over in the different parts of the code, for example the reading of configuration files, which each programmer does in their own way. How about some generalized configuration code module? Maybe this question is better for the dev list. Chaos seems to be the Asterisk way. :-) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] softhangup() problem
I have been trying to test out softhangup(). Every time I use it in a macro, it doesn't seem to hang up any call/s on the trunk. I have used: exten = s,1,SoftHangup(SIP/trunk-sx) exten = s,1,SoftHangup(SIP/trunk-sx|a) exten = s,1,SoftHangup(SIP/trunk-sx-1) exten = s,1,SoftHangup(SIP/trunk-sx-1|a) http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SoftHangup I don't know if I'm getting a option wrong or miss understanding its use, any help would be great. I'm a bit worried cause I've seen plenty of examples of e911 dial plans which use it... Thanks -- Shaun ___ --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] cmd DIAL - Who picked up the call?
Koopmann, Jan-Peter wrote:
On Friday, July 28, 2006 3:12 PM Kai Ober wrote:
What about DIAL ( |M(macro-name))
and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I would love to see the called
MSN in the port-field something like Zap/10-43 if MSN 43 was called... :-) That
would help enourmously.
Zap/10-43 would indicate that this is the 43rd call (call waiting) on
channel 10. Obviously this would have to be removed to do it the way
you want.
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[asterisk-users] SER local as an Asterisk Trunk
Hi, Would just like to ask, I have an SER SIP Proxy and I setup an Asterisk, i used an SER local as a trunk for the Asterisk. When the Asterisk box register to SER it will have this URI sip:[EMAIL PROTECTED], instead of sip:[EMAIL PROTECTED] Anyone has encountered this problem? Because I'm checking the From part, and s is not a valid extension number so it will deny it calling to the gateway. TIA Regards Nhadie ___ --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] Re: [Serusers] SER local as an Asterisk Trunk
Hi Last week i was working on the same i had same problem later after struggling lot, i have found solution by trying some options iam able to succeeded for the same may be this config should help you sip_additional.cfg register=account:[EMAIL PROTECTED]/account [ser][EMAIL PROTECTED]type=peersendrpid=yessecret=passwordqualify=yesnat=yesinsecure=veryhost= ser.serdomain.comfromuser=accountnamefromdomain=serdomain.comauthuser=accountname ram On 8/2/06, Nhadie Ramos [EMAIL PROTECTED] wrote: Hi,Would just like to ask, I have an SER SIP Proxy and I setup an Asterisk,i used an SER local as a trunk for the Asterisk. When the Asterisk box register to SER it will have this URIsip:[EMAIL PROTECTED], instead of sip:[EMAIL PROTECTED].Anyone has encountered this problem? Because I'm checking the From part, and s is not a valid extension number so it will deny it calling tothe gateway.TIARegardsNhadie___Serusers mailing list [EMAIL PROTECTED]http://lists.iptel.org/mailman/listinfo/serusers ___ --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
