[asterisk-users] One way voice after call transfer (bugs 9305, 13120)
Hello, I am having an issue here that after an attended call transfer there is no audio on one way; the problem is caused by Asterisk sending two INVITE messages without waiting for an ack for the first one. The issue has been reported on bug 9305, has been fixed and the fix is now included inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug (13120). This bug sits there for over a week with no reponse... Has anyone else noticed this behaviour? Any idea what I can do? My users are angry on me... Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changinf Polycom-501 config server from remote?
Hello, Our Polycom-501 phones are set to retreive their config for the server by a static configuation defined at the phones (boot servers). Is there any way to change it remotely? I found no relevant field in the internal WEB browser, nor anything in the configuration files (sip.conf and phone1.conf). Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] imap voicemail
Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where did I go wrong? Note that port 993 is IMAPS and I think you are telling it to use IMAP. Note also that Asterisk can get hung once in a while when using IMAP storage. Try netstat -an | grep 5060 and see whether you have a queue building there. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring back when free?
Has anyone here implemented Ring back when free in Asterisk?
Here is what I do; the dialplan enclosed is in AEL2 format, but you can get the
idea.
When a call is originated I save the called and callee numbers in a database.
If the user gets busy he/she hangs up and dial *41. I then retreive the last
number they dialled and place a flag in the database that someone is camping on
it. The H extension checks this flag and if found generates a .call file.
First, I have a macro to save the for each extension who is the last they
called and the last who called them:
// Save the calling and called numbers in To and From and in the database so
// they can be used by *41 and *42. This way the h extension can acecss this
// database for all destinations.
macro Save_From_to ( ) {
// To and From are used in the dialplan, since we might change ${EXTEN}
Set(_To=${MACRO_EXTEN});
Set(_From=${CALLERID(num)});
NoOp(= ${From} - ${To});
// Save them in database for later use.
// Save the caller number at the called extension for *42 usage.
Set(DB(${To}/LastCaller)=${From});
// Where we called for *41
Set(DB(${From}/LastCalled)=${To});
};
This macro is called at the beginning of the normal dialplan.
Now, the *41 which registers the camp-on using the data saved above:
// *41: Camp on the last extension dialled
*41 = {
Set(tmp=${DB(${CALLERID(num)}/LastCalled)});
// Save it so when the other side hangs it will see it and dial us.
Set(DB(${tmp}/CallBack)=${CALLERID(num)});
// Say the number to caller so he can verify...
SayDigits(${tmp});
Hangup();
};
And now the H extension for handling it:
// The Hangup extension which is called when the call is hanged. See whether
// we have some waiting callback waiting on this extension.
h = {
ResetCDR(w);// To make the CDR correct.
NoOp(${From});
// We have to check the two sides of the call: Those who camp on the calling
and
// those who camp on the called.
Set(tmp=${DB(${From}/CallBack)}); // The calling.
if(${tmp} != ) {// Something is there.
DBdel(${From}/CallBack); // And delete it...
// Create the callfile and then move it to the spool directory to make the
call.
System(echo Channel: SIP/${tmp}
/tmp/test.tmp${From})
;
System(echo WaitTime: 20 /tmp/test.tmp${From});
System(echo Extension: ${From}
/tmp/test.tmp${From});
System(echo CallerID: Callback \\\${tmp}\\\
/tmp/te
st.tmp${From});
System(mv /tmp/test.tmp${From}
/var/spool/asterisk/outgo
ing/);
};
Set(tmp=${DB(${To}/CallBack)}); // The called
if(${tmp} != ) {// Something is there
DBdel(${To}/CallBack); // And delete it...
// Create the callfile and then move it to the spool directory to make the
call.
System(echo Channel: SIP/${tmp} /tmp/test.tmp${To});
System(echo WaitTime: 20 /tmp/test.tmp${To});
System(echo Extension: ${To} /tmp/test.tmp${To});
System(echo CallerID: Callback \\\${tmp}\\\
/tmp/te
st.tmp${To});
System(mv /tmp/test.tmp${To}
/var/spool/asterisk/outgoin
g/);
};
};
Good luck! __Yehavi:
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Re: [asterisk-users] BLF and Snom phones
Sorry for the hijack, but I was wondering if I could be pointed at how
to get BLF and pickup working with Polycoms? I can use their Buddy
option to get a basic BLF system working, but that does not understand
the ringing state, and does not allow call pickup.
This is what I do (it works most of the times...). Our extension numbers are 5
digits, and I'll take as example making a BLF for extension 89444:
1. In the buddies list, instead of setting the BLF to be 89444, set it to be
*789444 (i.e preppend *7 to the number).
2. change the Hint to be (I am using AEL):
hint(Sip/89444) *789444 = NoOp();
This allows the phone to send subsciptions for *789444 which is translated
to the real extension's state.
3. Define the extension *7x:
// Directed pickup by *7. When we want a line key on the phone to be both
// speed dial and direct pickup we program it to dial *7num; We strip off
// the *7 and then check the destination status; if it is ringing - pick it;
// if not - dial it.
// When picking up - Set the callerID to be the orignaly calling station.
_*7X = {
NoOp();
Set(_To=${EXTEN:-5});
Set(_From=${CALLERID(num)});
if(${DEVSTATE(SIP/${To})} == RINGING) {
Set(CALLEDID(all)=${DB(${To}/CallingID)});
Pickup([EMAIL PROTECTED]);
Hangup();
} else {
goto huji-local|_8|StartLocal;
};
}
If the destination is ringing, pick it up. If not, strip the *7 and dial it.
I am using To and From since I need them for other places.
4. At your regular dialplan set the PICKUPMARK to the extension number:
Set(_PICKUPMARK=${EXTEN});
good luck, __Yehavi:
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Re: [asterisk-users] Realtime replication!!!!!
what do you think is the best way to replicate an asterisk mysql realtime database , i'm setting up a cluster , and i obviously need to ensure the database high availability i've been reading about DRBD and the internal replication functionality of mysql , but i dont really know if its the proper solution, i'm looking for advices on how to replicate the mysql realtime database so then when the main server is down , the second replace it . I won't call it best way but I am using two-way master-slave repliaction of MySQL. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web
I am using version 2.2.0. __Yehavi: Date: Thu, 6 Mar 2008 15:01:26 +1100 From: Lee, John (Sydney) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass As far as I recall it can be done from the config file only. Here is the relevant line from sip.cfg: device device.set=1 device.auth.localAdminPassword.set=1 device.auth.loc alAdminPassword=YOUR-PASSWORD-HERE / What sip release are you referring to? I am looking at sip 1.6.x and sip.cfg only allows you to set the length of the user and admin passwords. You cannot set the password in cfg. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to restrict a Polycom from receiving unauthorized calls
Hello, I've found that my Polycom-501 accepts INVITES from any server in the world... I would like to restrict it to accept calls only from the servers listed in its config file, but I cannot find anything in the documentation. Any idea? Thanks, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass via Web
I could be mistaken, but I think this has to be done physically from the phone. I don't think you can do this with central provisioning or from the web interface. As far as I recall it can be done from the config file only. Here is the relevant line from sip.cfg: device device.set=1 device.auth.localAdminPassword.set=1 device.auth.loc alAdminPassword=YOUR-PASSWORD-HERE / regards, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing
If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I See the discussion a few days ago. The Asterisk server saves the value of SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL. Regards, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing SIP extensions.
Hello,
Here is how I do this. The prerequisits are:
- MySQL to hold the extensions realtime database. MySQL is synchronized
among all servers using the Master/slave replication model.
- The phones are spread by some external algorithm over the Asterisk servers
(statefull load balancer, statically defined in the config file of the
phone, etc.).
The idea is to locate on which server the destination phone is registered and
redirect the call to it. For this:
/etc/asterisk/asterisk.conf has the parameter sysname set to its IP address
(you can use also a DNS name, but I want to be independent of name
resolution). This causes the server to set the field regserver to be
saved in the MySQL database to the IP address of the server.
/etc/asterisk/extensions: The logic to check whether the value of regsever
is different from sysname and if so - redirect the call. The code
fragments are (I am using AEL):
To get the regserver from the database:
MYSQL(Query resID ${connid} SELECT regserver from sip_users where
name='${EXTEN}');
MYSQL(Fetch FetchId ${resID} RegServer);
MYSQL(Clear ${resID});
so now RegServer contains the server where the phone is registered. Next:
if((${DEVSTATE(SIP/${EXTEN})} == UNAVAILABLE) ||
(${DEVSTATE(SIP/${EXTEN})} == INVALID)) {
if(${SYSTEMNAME} != ${RegServer}) {
Transfer(SIP/[EMAIL PROTECTED]);
return;
};
};
I check for the device state so in case the phone has double registration
(primary and backup server) it will be processed localy.
Regards, __Yehavi:
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Re: [asterisk-users] load balancing SIP extensions
What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there ... I've tried doing something similar and came with two options. The common to them is that I use MySQL for realtime extensions, and set systemname parameter to the IP address of the server where the phone registers. When a call arrives I check whether the REGSERVER coloumn is the same as the local server or not. If not, then there are two options: - Pass the call via IAX to the other servers; this makes both server process the call and the audio. - Send a refer message to the caller to contact the other server. I had this working in the lab but not in production yet. If you want the dialplan code for this then email me. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
Hello, I've one nokia E65 that works very well with my asterisk box. The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Thanks, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco SIP Gateway
Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco 1751 with VIC-2BRI-NT/TE talking SIP to Asterisk, much like like an SPA3K in the analogue world. Any success stories? I am using a Cisco router as a PRI gateway with SIP. I have no experience with BRI, but it should work as long as you have DSP cards for voice processing (the BRI card can do only data calls, and DSPs are needed for voice). __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server
I'm trying to test IMAP in 1.4.17 and it appears to be not working. I've compiled imap-2007 with the following on a CentOS 5 box: make slx EXTRACFLAGS=-I/usr/include/openssl -fPIC and I've configured and compiled asterisk with the following: ./configure --with-imap=/usr/local/src/imap-2007 And now in make menuselect you have to go to voicemail options and set IMAP support to on. Here's my voicemail.conf: [general] imapserver=localhost imapfolder=Inbox ;pollmailboxes=yes ;pollfreq=30 imapflags=notls authuser=asttest expungeonhangup=yes authpassword=whatever I had to enable pollmailboxes in order to update MWI. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable call waiting by default
I've connected some analogic phone to some fxs modules on an analogic card. I want to disable by default the call waiting sound. I know that dialing *70 before to call the call waiting is disabled until the next call, but isn't there a setting or a dialplan command to set up this automatically? If you mean that there is no waiting call then use DEV_STATE function to see whether the called extension is in a call; if so - call the Busy() application. If you need more details then search in the lists's history - I;ve posted a code fragment about a month ago. __yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b2bua
On Friday 04 January 2008 16:45:00 ameel wrote: Is there a way to disable the b2bua feature in asterisk. I would like asterisk to work as a sip server and not be involved in the RTP path between phones. No. And by the way, b2bua is not a feature. It's is literally what Asterisk is. If I understand correctly the question the RTP can be established directly between the phones and does not have to pass through the Asterisk PBX. Just set CANREINVITE=YES and use the same CODECs for both phones. __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enable eyeBeam to accept only one call
Hello
I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
when they are still in one call (because eyeBeam has lots of channels).
I was using X-Lite (with 3 channels) and Asterisk never sent the client
a second call.
How can I force Asterisk (or eyeBeam) just to send one call at each time.
Is this a configuration I need to do in eyeBeam or Asterisk?
Please contact Eyebeam and request this feature - The more people who
ask for it the better. I have requested it previously to no avail.
It is possible to emulate this in asterisk using inuse counting, but I
still get the occasional off-by-one that requires an asterisk restart
to bring a phone back to life.
I am doing that on 1.4.15 and it works fine so far. The code fragment is:
if(${DEVSTATE(SIP/${To})} != NOT_INUSE) {
Busy();
};
Note that for function DEVSTATE you have to install the relevant function. See
http://www.asterisk.org/node/48360
Regards, __Yehavi:
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Re: [asterisk-users] retrieve last number dialled
What is the easiest (simplest) way to do this?
I do it in two steps: Save the dialled number in Asterisk DB and have a
special extension (*41) which redials it.
Here is the abstract from the dialplan where I save it:
Set(_To=${EXTEN}); // Save the original extension dialled.
Set(_From=${CALLERID(num)}); // Save the caller.
Set(DB(${To}/LastCaller)=${From});
Set(DB(${From}/LastCalled)=${To});
and then the special extension:
*41 = {
Set(tmp=${DB(${CALLERID(num)}/LastCalled)});
SayDigits(${tmp});
// Save it so when the other side hangs it will see it and dial us.
Dial(SIP/${tmp});
};
Regards, __Yehavi:
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[asterisk-users] Finding the status of an extension
Hello, I would like to check whether an extension is busy or not before calling the Dial() application to it (for example - to play a Busy if it is on conversation). How do I check it? In the trunk version there was a function DEVSTATE(SIP/123), however it does not exist on version 1.4.13... What is the equivalent of it? Thanks, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need T1 crossover cable?
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? As people answered here you need a crossed cable; Note that T1/E1 cables are different than Ethernet. T1/E1 uses pins 1,2,4,5 (while Ethernet uses 1,2,3,6). A crossed T1 cable is: 1 4 2 5 4 1 5 2 Regards, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phones and corporate phone directory
Hello, A few days ago I've posted two questions about Polycom phones: How to access corporate phone directory from the phone and how to use a conference server with it. After I got zero responses I tried openning a support call in Polycom's site. Here are the replies I got from them: - Corporate directory: They are thinking about it, probably will use LDAP. I was asked to open a feature request to which I got no response yet. - conference server: I asked it because there is a a parameter to set an external conference server, and when I tried it only the initiator was transfered to the server; the other parties were left in hold state. Polycom replied that it is not supported and only the phone's internal conference feature should be used. Regards, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unlocking Cisco 7921
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For the purpose of the pilot (i.e. low investment) I want to configure the phones from the keypad. Each phone shows settings locked! whenever I try to edit the network profiles. I can't seem to unlock them! Hopefully there is a secret button combination...I would hate to have to go to a Cisco Unified CallManager just to unlock a few phones... On other Cisco modemls the sequence to unlock them (as long as you did not set a password) is **# (star, star, pound sign). I hope it works on them as well. __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP usage with Asterisk
Yehavi Bourvine +972-8-9489444 wrote: In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. There have been _many_ IMAP related fixes sine 1.4.6. Please try the latest version, 1.4.13, instead. -- Russell Bryant Sorry, I had a typing mistake - I meant 1.4.13... Anyway, I tried it at the lab today and it works ok. There is one minor problem when parsing the IMAP headers (files a bug at ID 11043). I'll be able to move it to the production system only next week. Thanks, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about Plycom phones and Asterisk
Hello, I have two issues which I would like to know whether someone has an answer to them: 1. Our institute has over 8,000 phone numbers and I would like to allow people to search it from the phone. I am willing to write some XHTML scripts to run through the microbrowser, but I cannot find at the documentation what format it should be in order for the phone to understand the output and allow dialling of the numbers returned. 2. Polycom supports external conference server via the parameter voIpProt.SIP.conference.address. Any idea how should I include in it the conference room number (so I can pass it to MeetMe() application)? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco
Isee no one has answered you so far, so I'll try: I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then go to the VOIP serviece provider network ip-phone -asterisk-cisco-voip netwrok how can i do that , whats asterisk config file ( H.323, sip and extensions.conf ) and how is cisco configuration for this setup? Thanks Giti I've done this with Cisco-2,811 (the exact model should not matter) and with SIP (H.323 should be quite similar). On Asterisk you have to add a rule in the dialplan to direct the outgoing calls to the Cisco. In the most simple way it should be something like: Dial([EMAIL PROTECTED]) On the Cisco gateway you should have a dial-peer which calls the PSTN. For example: dial-peer voice 2 pots destination-pattern 0T progress_ind setup enable 3 no digit-strip direct-inward-dial port 0/0/0:15 (this will direct any call to 0xxx to voice card 0/0/0:15). If you need a more detailed config please contact me off-list. Regards, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP usage with Asterisk
Hello, I tried a few months ago to use IMAP with Asterisk; I used either 1.4 or the latest SVN at that time (sorry, don't remember). After a day I had to remove it since Asterisk crashed, mostly in the IMAP client code (the code of UW IMAP). My users wants IMAP back (they loved it) but not in the price of crash... I could not reproduce the crashes at the lab. They only occour on the live system with users. I suggest to write the IMAP client code by the Asterisk developers and not depend on external code. In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Difference between trunk and released versions
Hello, Up to a while ago I thought that the released versions are checkpoints of the trunk versions; however, now I understand they are not, as I see differences between the two trains. So, what is the relation between them? Examples for differences: - When the language is different than Engligh the trunk version is reading numbers from /var/lib/asterisk/sounds/Lang-Name/digits while the release version is using /var/lib/asterisk/sounds/digits/Lang-Name - MAILBOX_EXISTS function is replaced with MailboxExists application. - External IVR has no way to exit from the program under the release version... The documentation is correct with the trunk version. Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between trunk and released versions
Hello Sean, Does this clear things up? Yes! Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.conf vs. AEL
Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] online active call watching
try the astman command. __Yehavi: ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Show Callee name on Display
SInce no one else has brought this up, just thought I'd let you know that it is being worked on... http://bugs.digium.com/view.php?id=8824 And it works - I am using it for months already. Note that not all phones support it. Cisco and Policom supports it, while Snom does not. __Yehavi: ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExternalIVR() broken in 1.4.9?
Hello, After upgrading to 1.4.9 the above function does not work anymore; it claims that child went away while the child is probably not born at all... Before I open a bug on it, anyone has a clue? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for a USB phone handset or headset
Hello, I would like people to use soft phone, but they are used to have the standard phone handset in their hands... Is there a USB handset or a handset that connects to the audio card? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TCP-UDP SIP proxy?
Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi: ___ --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] SLA with SIP-only environment
Hello, All the examples of SLA talk about Zap channels from one side and SIP on the other side, while my system is a pure SIP one. I would like to have two phones having extensions 1 2 defined on them, and when someone calls extension 1 it rings on both, each one can see its status, and when one station puts line 1 on hold the other one can pick it. Is it possible at all? If so, can someone give the relevant fragments for sla.conf, sip.conf and extensions.conf? Thanks! __Yehavi: ___ --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: Call waiting tone
Hello,
A few days ago I've asked about the ability to play a stuttered ringing
tone when the called party is already on the phone. I've found a partial
solution for it.
To describe again the problem: When a user is on a call and someone else
calls him, the caller does not know that the called party is on the phone
(while the called party wants to know that someone else is calling him/her and
not just play busy). On our public PSTN the caller is notified by a stuttered
ringing tone (thus he can decide whether to wait or hangup and call later).
What's I've done is that when the called party is on the phone a short message
is sent to the caller (may be also a recording of the stuttered ringing) and
then the call is passed. Here is the code fragment:
exten = _806XX,n,Set(Status=${DEVSTATE(SIP/${EXTEN})}) ; Get his status
exten = _806XX,n,GotoIf($[${Status} == NOT_INUSE]?OK:WAITING_CALL)
exten = _806XX,n(WAITING_CALL),Playback(waiting-call)
exten = _806XX,n(OK),Dial(SIP/${EXTEN}${aEXTEN},20,) ; Dial the phone for 20
Hope it helps someone...
Regards, __Yehavi:
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[asterisk-users] Call waiting tone when calling a busy station?
Hello, When dialling a SIP phone which is already in a call the caller hears a regular ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi: ___ --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] Call waiting tone when calling a busy station?
this is a SIP phone configuration issue. You should tell the UAC to not accept a second call while the line is engaged (look for a 'Call Waiting' option in the configuration of the UAC) The UAC will send back a 486 Busy Here error code and the calling party will get a busy signal from asterisk This is not what I meant. I want the called party to get a sign of a waiting call and answer it if he/she wants. I want the caller to know that he on a waiting call (here it is customary to play a stuttered ring tone). in short - can I signal in the 183 ringing packet that this is a second call? Thanks! __Yehavi: ___ --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] Called party identification - where to take called name?
Hello, I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Thanks, __Yehavi: ___ --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] Called party identification - where to takecalledname?
Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Asnwering myself: I am using realtime extensions, so I've added call to MYSQL() application to get the called user callerid field. __Yehavi: ___ --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 application in dial plan
Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? Thanks, __Yehavi: ___ --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] [OT] How to find out line that you are on from Bezeq
Does anyone know the code to dial in Israel on Bezeq to get back the number of the line that you are on ? Dial 110 and it will tell you your number. __Yehavi: ___ --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] SLA with SIP only configuration
Hello, I would like to understand how SLA is working, but all the examples are using trunk of ZAP type and stations of SIP. However, in my case the stations are SIP and the outgoing connection to PSTN is SIP also (link to a Cisco gateway). can anyone send me a simple configuration of how to make SLA between two SIP phones without dependency of anything else? It seems I lack some basic understanding of thios feature and maybe such an example can help. Thanks! __Yehavi: ___ --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] Connection between Asterisk - Cisco 2851
Hello from Greece. I have an assignment related to Cisco. Specifically, i have to connect an asterisk server (VoIP) to a Cisco Router 2851 in a way that the router will handle some calls. I have not found any manuals that would be helpful to me. On the other hand, i found some manuals for connection between asterisk and Cisco Call Managers. I would like to know if the call manager is a part of the router or is an extra element. I would appriciate if you could help me. I'll just add one more thing: Your Cisco must have DSPs in order to allow the E1 to have voice calls (on 2,811 it is called PVDM). Without the DSPs your E1 can do only data calls... __Yehavi: ___ --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] Cisco GW, PRI CallerID Name
Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway via SIP to *? I've seen a few people ask and a few people that say it should work, but I've never seen an actual working config. I have it working, but it depends on the specific configuration. I have it working via PRI/Q.sig to a Nortel TX-1. The key issue here is that the name can be sent in two ways: via the Q.931 and via Q.sig. Nortel supports only Q.sig, and you need to enable it on the Cisco. If this is your case then tell me and I'll send the config. __Yehavi: ___ --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] SIP - Automatic Redial on No Answer
Here are the relevant parts from extensions.conf file. It works only for local
extensions whose number id 806xx. Note one thing: When you use the H extension
the generated CDR is wrong - the destination extension is H and not the
original number. I've done some small code change in Asterisk and has to file a
bug about it.
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
;;; First, save the caller and called numbers for *41 and *42 features.
exten = _806XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled.
exten = _806XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.
; Save the caller number at the called extension for *42 usage.
exten = _806XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _806XX,n,Set(DB(${From}/LastCalled)=${To})
; See whether this extension needs to ring at others as well. To define such
; an extension do inside the command prompt of Asterisk:
; database set EXT-NUMBER MoreLinesToRing SIP/firt-lineSIP/Second-line
exten = _806XX,n,Set(aEXTEN=${DB(${EXTEN}/MoreLinesToRing)})
; Now dial the extension.
exten = _806XX,n,Dial(SIP/${EXTEN}${aEXTEN},20,) ; Dial the phone for 20
seconds.
; No answer or busy
exten = _806XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure
mode
exten = _806XX,n,Hangup() ; Just to be sure...
; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(${To}|u) ; Has mailbox - send the call to
there
; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(${To}|b) ; Has mailbox - send the call to there
; Congestion: Handle the same a busy.
exten = s-CONGESTION,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-CONGESTION,n,Busy() ; No maibox = play busy.
exten = s-CONGESTION,102,VoiceMail(${To}|b) ; Has mailbox - send the call to
there
; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);
; Called here when the call is successfull and the user hanged the phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make the call.
exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/)
; No one is waiting for us...
exten = h,103,NoOp(Nothing to call)
; The activation codes for the above code.
; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)
; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
Good luck, __Yehavi:
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Re: [asterisk-users] SIP - Automatic Redial on No Answer
Hello, I've wrote a dialplan script which uses the H extension to do something similar to what you want. In general it uses the internal ASTDB for this: - When there is no answer (or busy) the caller hangs up, initiate a new call with some special code (*41 is used here by the public carrier, so I am using it also). Asterisk registers the data in its DB. - When the user disconnects the H extension is called. It then looks in ASTDB to see whether there is a user camoing on this extension. If so, a call file is created and Asterisk initiates the call. If this is what you need please tell me and I'll post the code on Thursday. I've already posted it in the past so you might search the archives in the meantime. __Yehavi: ___ --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] Conference server (or how to make a call with more than 3 u
On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi: ___ --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] Conference server (or how to make a call withmore than 3 u
Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I have a few Snom phones here - people do not like them... Thanks, __Yehavi: ___ --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] Conference server (or how to make a call with more than 3 u
Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee Possible, but looks too much cumbersome... However, that's a nice idea. Thanks! __Yehavi: ___ --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] Conference server (or how to make a call with more than 3 users)
Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Thanks, __Yehavi: ___ --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 and Cisco PRI gateway config
That was exactly what I meant. Your setup is : Nortel --- Cisco --- Asterisk What I was thinking about is: Nortel --- Asterisk1 --Asterisk2 In previous case, your are using Cisco's QSIG features. In the latter one, you could use Asterisk QSIG features. I was asking because, I was wondering how Cisco and Asterisk QSIG features compare. Sorry, I have no idea... __Yehavi: ___ --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] SetCIDNum is not available on 1.4svn
Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... Thanks! __Yehavi: ___ --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 and Cisco PRI gateway config
do you have some info, about what platforms supports qsig decode? I found, that first supported in 12.4.(9)T, but I don't know, if only on 28xx or also in older routers (like 3660) with NM-HDV-E1... I assume that it is a hardware independent but I do not have the hardware to test... I see refernce to this command in 12.3T documentation, so it should be inside the normal 12.4 version. I moved to 12.4T due to TAC request before I tried the above command, so I did not test it on 12.4. __yehavi: ___ --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 and Cisco PRI gateway config
Do you think it could have been done with another T1/E1Asterisk box between the Nortel PBX and the other Asterisk server ? Sorry, I do not understand exactly what you are asking. Do you mean using an Asterisk with PRI card instead of Cisco? If so, I have no experience with this. Which features would you then loose or gain, given current status of QSIG support in Asterisk ? In my case the Cisco did all the Q.sig work so Asterisk's Q.sig functionality was not used. __Yehavi: ___ --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] Asterisk and Cisco PRI gateway config
Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined to send and receive names via Q.sig. The definition fragments on Cisco are: isdn switch-type primary-qsig (so it will use Q.sig signalling). ... voice service voip qsig decode(This sends names out via Q.sig) fax protocol pass-through g711alaw sip controller E1 0/0/0 pri-group timeslots 1-31 ... interface Serial0/0/0:15 (This is for E1 PRI). no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling (This receives names via Q.sig) isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable Anc the rest is quite standard. Regards, __Yehavi: ___ --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 and Cisco PRI gateway config
interesting! so it means, that you can now see caller id names between sip phones connected to asterisk and phones connected to pbx? Yes! __Yehavi: ___ --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] Lastest SVN (1.4) and realtime call limit
Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left in use and could not receive new calls. - After seeing reference to similar problem on this list I;ve downloaded today the latest SVN source code and installed it. The problem is that it shows the call limit as 0 and not as 1. Any idea? Thanks, __yehavi: ___ --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] 7912 phones loosing registration
I had a similar problem with Asterisk-1.4; After digging slightly I've found that sterisk crashed (safe_asterisk restarted it and I did not notice) and this caused the phones to lose registrations until the next expiration... I am not sure this is your case, but try TCPDUMP of port 5060 from these phones. It might give a clue. __Yehavi: Date: Thu, 15 Feb 2007 13:57:47 -0500 From: Jerry Geis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] 7912 phones loosing registration I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry ___ --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] Cisco Router for supply a connection from PABX to Asterisk
anyone know if they have a solution in Cisco for: 1- Connect old PABX (with BRI or PRI) to a cisco router 2- Connect this cisco router in SIP to a Asterisk Server I am search if cisco can this and what is the modele for this I am using a Cisco to connect Asterisk via PRI to our Nortel TX-1. The Cisco is a voice bundle of 2,811 + E1 + PDLM card. Note that you need PDLMs as the same number as the PRI channels you are going to define (i.e. 32 PDLMs for each PRI). I am controlling the Cisco via SIP; it works, but a few problems: - Only basic connectivity. No additional features (like names) as the Cisco supports them only via MGCP (in MGCP is passes all the Q.sig signals to the PBX - Asterisk in this case - and it should do all the handling, but I did not find how to do it with Asterisk). - Cisco has no authentication mechanism, so anyone which has access to port 5060 of it can generate calls. Asterisk is not better when it concerned in such sutiations... Regards, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM phones stay in use after transfer
Hello, Before I dig into the SIP traces maybe someone has a clue: We have a few Snom phones, and the call-limit on the extension associated with them is set to 1 (so the other Snom phones can blink the relevant LED). When I call a SNOM phone (300 with the latest firmware) and the call is answered the phone is in in use state (verbose printing from Asterisk). When I hit the transfer button it gets into hold state in Asterisk. I dial the new extension and the call seems to be transferred (actually it goes to hell...). Anyway, the Snom is now in left in hold state forever, and Asterisk won't accept new calls to it since it has max calls of 1... Similar thing happens with attended transfer (putting the call on hold, calling the other party and either disconnecting or pressing Transfer). Any idea? Thanks! __Yehavi: BTW, any idea when 1.4.1 is scheduled to be released? I need the SLA feature desperately... ___ --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] Cisco PRI gateway with MGCP control
Hello, Anyone managed to control a Cisco voice gateway (2,811 in my case) using MGCP? I cannot make the PRI going on-line (while with SIP I can). If you ask why I want to use MGCP and not SIP: it is because Cisco uses different Q.sig signalling when you manage it with different protocols, and I need the other Q.sig... Thanks, __Yehavi: ___ --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] OT - Cisco 7960 functionality
The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want to push the answer softkey after they pickup the handset in order to answer a call. Doesn;t it answers when you pick up the handset? Here it does so... The users want the transfer softkey on the screen while on a call. Currently it is acessable via the more softkey. I've asked Cisco whether all the above can be done and got a negative reply. __Yehavi: ___ --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] PRI/Q.sig between Cisco Nortel
Hello, I am using a Cisco-2,811 as a gateway between the Asterisk PBX and our Nortel TX-1 university's PBX. It is working but no names are exchanged. From the debug mode I see that the Cisco sends the display name (which does not appear on the Nortel's phones) and the Nortel does not bother to send it at all. I recall that when I had a pilot with Cisco CCM two years ago we had to set the siganlling to ESGF on the Nortel and use MGCP on Cisco (since MGCP protocol forces the Cisco to use ESGF signalling). We could not use H.323 as it forces the Cisco to use ISGF. I suspect that SIP is the same, but setting ISGF signalling on Nortel doesn't help. Anyone had some luck with this configuration? Thanks, __Yehavi: ___ --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] Callback/ringback
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled.
exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.
; Save the caller number at the called extension for *42 usage.
exten = _999XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _999XX,n,Set(DB(${From}/LastCalled)=${To})
; Now dial the extension.
exten = _999XX,n,Dial(SIP/${EXTEN},20,) ; Dial the phone for 20 seconds.
; No answer or busy
exten = _999XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure
mode
exten = _999XX,n,Hangup() ; Just to be sure...
; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call to there
; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to there
; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);
; Called here when the call is successfull and the user hanged the phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make the call.
exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/)
exten = h,103,NoOp(Nothing to call)
; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)
; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
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[asterisk-users] Disallowing unauthorized calls to Cisco Polycom phones
Hello, I would like the IP phones to not accespt SIP requests (like INVITE) from any device other than its proxy. Snom phones ignore this while Cisco Polycom accepts the call. Any idea what to do to disable it? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] real life example of SLA definition
Hello,
I am looking for a real life example of using SLA lines under Asterisk.
I'll describe my environment and would like to know how I define it in
Asterisk (version 1.4 final).
Suppose I have two multi lines phones. The first phone has extension 1
assigned to it, and the second phone has extension 2 assigned to it. Now, I
want extension 3 to be available on both phones as additional shared extension.
What shall I define on the phone? (define extension 3 on both doesn't work as
only one can register with it). What should sla.conf file have? Do I have to
change extensions.conf? (To make it simple let's assume that it contains only
Dial(SIP/${EXTEN}) as the dialplan).
Thanks! __Yehavi:
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Re: [asterisk-users] real life example of SLA definition
You say you can't put 'extension 3' on both phones because only one can register to it, but that is incorrect. Extensions and SIP devices are not the the same thing; extensions exist in the dialplan, and SIP devices are in the SIP channel driver. Create two SIP friends for the two phones to use for their 'extension 3' line appearance (call them 'phone1-ext3' and 'phone2-ext3' or something), get them both to register, and then dial both devices from the dialplan when extension 3 is called. OK, but how do I make that when extension 3 is being used on one phone the other one can see its status, and when it is placed on hold then it can be picked-up on the other phone by just hitting the line button? I am trying to emulate the current phone system behaviour, as this is what the users want... Thanks! __Yehavi: ___ --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
