similar to: SetVar - bellcode and cisco phone

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

Hello, I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the /etc/asterisk/capi.conf I have in /etc/asterisk/capi.conf 's section "[interfaces]" the following directive context=isdn and the following directive in /etc/asterisk/extensions.conf in

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

it is possible to use musiconhold. i added exten => s,5,SetMusicOnHold,default exten => s,7,dial,SIP/michael&SIP/frank|15|m to the extension conf. the logfile looks good, i think -- Called michael -- Called joern -- Started music on hold, class 'default', on CAPI[contr1/9999] -- SIP/192.168.100.150:5060 is ringing -- Stopped music on hold on CAPI[contr1/9999] --

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

Hello all, I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA) on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one B channel. But when I open a second B chan, the sound is choppy, with too long gaps, and the CPU load is too high (~50%). On the Asterisk's console I get these messages: -- Executing Dial("H323:4478",

Hi, I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi (Fritz PCI). When I call a national number, I?m hearing the ringtone when the called party is ringing but when I call an international number, I don?t hear the ringtone and I?ve a silence until the called party answers. Both call are going through the same extension. Here?s 2 log files, one with a national number and

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

A call comes in I set the distinctive ring by setting variable ALERT_INFO then dial a SIP channel. The channel is answered, but then the user forwards the call to another SIP channel. ALERT_INFO is still set. How can I clear the ALERT_INFO variable after the SIP channel is answered so that when the call is forwarded the ring goes back to "normal"?

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

Hi All - I'm trying to implement the auto-answer config from the wiki, but the result for me is that the phone just rings as normal. I'm running firmware version 1.4.1 on an IP500. I've added the following to my sip.cfg: <alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"/> and this to my ipmid.cfg

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making