[Asterisk-Users] Re: Voicemailmain automatic extension detection?
Hello Mason. It's easier than you might think. Here's what we do to
achieve the same effect. Note that, depending on how many digits you store
in your voicemail configs, you may need to change the number of stripped
digits, etc.
We use 3-digit dialing for voicemail, and the mailbox number matches the
last 3 digits of the user's telephone number.
-Brian
;Voicemail access
exten = 299,1,Wait,1 ; Just to see if it's working...
exten = 299,2,Answer ; Answer the call
exten = 299,3,SubString(MBOX=${CALLERIDNUM}|-3|3)
exten = 299,4,Voicemailmain(${MBOX})
exten = 299,5,Hangup
On Oct 5, 4:04pm, [EMAIL PROTECTED] wrote:
} Is there a way I can have voice mail check calls coming from my internal
} users automatically get to the right extension, without having the user
} enter their extension?
}
} I'm thinking that I could have the local SPA boxes translate, or have
} each user live in a context where the extension in question exists
} uniquely per user, but both of these seem kludgey.
}
} Thanks in advance for clues!
}
} --
} Mason Loring Bliss [EMAIL PROTECTED] http://blisses.org/
} Anything can be impossible, given sufficient bureaucracy.
}
}
-- End of excerpt from [EMAIL PROTECTED]
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[Asterisk-Users] Re: Interrupting voicemail with *, dropping to a
I'd be curious about this as well. In Asterisk version 1.0.7, it can't possibly work, unless my C reading skill is completely broken, because the voicemail app isn't listening for a * but only for a # or a 0. That's also true of /app_voicemail.c/1.203/Thu Mar 10 19:33:15 2005//D2005.03.10.08.00.00 For those interested, I've created a patch to the app_voicemail.c file, which I've been using for about a year and a half, which drops you into voicemailmain if you hit * during the outgoing message. As far as I can tell, I still need this patch as of the latest CVS version. Anyone who has a working example and CVS dates to clearly identify when the feature went in should speak up, as it seems there are a number of us who would like the feature. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Broadvoice latest changes and still not working
In looking into this further, it appears that the problem is that ASterisk is not properly responding to the 401 request that comes back from BroadVoice. The code is there to do the right thing, and I can say that ASterisk does the right thing when a 407 response is received from a provider, but for some reason, Asterisk never gets the authentication response on the wire after Broadvoice asks for it. Someone wrote and said that they wer able to use BroadVoice with their SIP phone directly. Would it be possible for that person to send a SIP trace of the successful call so I could compare the two streams? I don't have a fix for this problem yet, but I hope to have one soon as my outgoing service is foobared until I can fix it. -thanks -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Broadvoice configuration changes for outbound calls
Hello. I'm not sure what's going on with the gentleman who is having trouble receiving inbound calls as of this weekend, but I can say that while inbound works for me, calling out through BroadVoice doesn't work at all. SIP traces show that when I send an invite request out to BroadVoice, they send back a 401 unauthorized message which includes a WWW-Authentication: header which ASterisk is supposed to use to send a reply proxy authentication response. The version of Asterisk I'm running, and have been running with BroadVoice for months claims that it sends an acknowledgement of the unauthorized message, then fails to send an authentication reply, instead claiming that authentication is impossible with BroadVoice. I suspect that there is a bug in the md5 hashing code on the version of Asterisk I'm running, and I'll be attempting to upgrade things, or sort out the bug soon. My point here is to let people know that they may be seeing different behaviors depending on what version of ASterisk code they're running. I'm running with CVS head as of 2003-12-18. I doubt many others are running code this old, but until this Saturday morning, it's worked flawlessly with every provider I've tried it with. Having said all that, I too am disappointed that BroadVoice has not seen fit to tell its users of this impending change. Instead, it worked on Friday night for me, all normal, and, voila! complete failure of outgoing calls on Saturday morning. Most disturbing. Hope that's somewhat helpful. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins
Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, Juki wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to } sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones } on two different PCs. My problem is that when one of the SIP phones logins } in, the other won't. } } My sip.conf has: } [101] } type=friend } host=dynamic } username=101 } secret=test } dtmfmode=rfc2833 } context=from-sip } mailbox=201 } callerid=101 2125 } nat=yes } } My extensions.conf has: } exten = 101,1,Dial(SIP/101,20,tr) } exten = 101,2,VoiceMail,u101 } exten = 101,102,VoiceMail,b101 } } My voicemail.conf has: } 101 = 2348,Emma, [EMAIL PROTECTED] } } Any ideas are most welcome. } } -- } Rgds, } Juki } } ___ } Asterisk-BSD mailing list } [EMAIL PROTECTED] } http://lists.digium.com/mailman/listinfo/asterisk-bsd -- End of excerpt from Juki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dtmfmode: inband question
Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like: [8315551212] type = friend ... dtmfmode = inband ... When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the other end? What I'm noticing is that if I call a pstn line using an entry like this through asterisk, and then press digits on the SIP phone connected to asterisk, I hear very short tones on the pstn line instead of the long tones I generate on the SIP phone. In addition, if I press digits too quickly on the SIP phone, where too quickly is not very fast at all, many digits are dropped entirely and do not make it to the pstn phone at all. It occurred to me that this might be a fixable problem in the Asterisk source code, but when I read the code itself, it is not clear to me who is generating these short dtmf bursts, and perhaps it is the fault of the SIP instrument, a Cisco 7960 running SIP image 6.2, it self. So, if anyone can explain to me where the DTMF tones are coming from when the dtmfmode is set to inband, I'd be most appreciative. -thanks -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
Hello. Here is what my extension which uses distinctive ring on a Cisco 7960 running V6.2 firmware looks like. Note that the distinctive ring tones are changes in cadence, rather than changes in ringing sounds on the 7960. Also, if you adjust the ringer volume wile the distinctive ring is sounding, the phone will revert to the non-distinctive ring cadence. -Brian } Hi } } Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? } } I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. } } Thanks in advance. } } P } ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
[Try this again...] Hello. Here is what my extension which uses distinctive ring on a Cisco 7960 running V6.2 firmware looks like. Note that the distinctive ring tones are changes in cadence, rather than changes in ringing sounds on the 7960. Also, if you adjust the ringer volume wile the distinctive ring is sounding, the phone will revert to the non-distinctive ring cadence. -Brian exten = 2135551212,1,setvar(ALERT_INFO=4) exten = 2135551212,2,Dial(SIP/100SIP/401SIP/403|20|tr) exten = 2135551212,3,Voicemail,u401 } Hi } } Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? } } I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. } } Thanks in advance. } } P } ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstreams randomly go busy with Asterisk?
Hello. I've seen this behavior. What happens is that the Grandstreams forget to continue registering with Asterisk after a while. I bet when you find this happening, that sip show peers doesn't show ext/ext ip address for the one that isn't working. You can work around the problem by explicitly telling Asterisk how to dial the GS by giving it an explicit IP address in its sip.conf extension entry. Alternatively, you can upgrade the Grandstream to a newer load of firmware. I'm running 1.0.4.68 on my HT286, and it seems to behave much better. I got my firmware load from: http://www.voiptalk.org/products/download/ They seem to have 1.0.4.63, and 1.0.5.0, but not 1.0.4.68 anymore. Hope that helps. -Brian I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro(SIP/24567-7856, dialphone|SIP/27654) in new stack -- Executing Dial(SIP/24567-7856, SIP/27654|10|tr) in new stack Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy at this time But dialing in the other direction (from the busy phone out) gives normal (good) results: -- Executing Macro(SIP/27654-6e2b, dialphone|SIP/24567) in new stack -- Executing Dial(SIP/27654-6e2b, SIP/24567|10|tr) in new stack -- Called 24567 I have noticed that when the problem is happening I see this: CLI sip show peers Name/usernameHost Mask Port Status 24567/24567 192.168.2.253 (D) 255.255.255.255 5060 Unmonitored 27654/27654 (Unspecified) (D) 255.255.255.255 0Unmonitored Rebooting the offending phone always fixes the problem for a while. After rebooting I see: CLI sip show peers Name/usernameHost Mask Port Status 24567/24567 192.168.2.253 (D) 255.255.255.255 5060 Unmonitored 27654/27654 192.168.2.254 (D) 255.255.255.255 5060 Unmonitored The BT101s are running 1.0.4.55. Asterisk is 0.9.0. Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: External access to voicemail
Hello. I have written a small patch to app_voicemail.c which provides the precise functionality Steve wants. I sent it to this list once, and got my subscription disabled for my trouble. so, if anyone's interested in it, it's about a 50 line diff file, which I'd be happy to mail anyone who writes and says they want it. If enough write, I'll post a URL on this list for it. If it's super popular, I'll figure out how to submit it as a feature request on the bug tracker. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users]: External access to voicemail
Hello steve. Here is a patch I wrote for app_voicemail.c which does
exactly as you describe. When the outgoing message is playing, if the
listener hits the * key, they're prompted for a mailbox and password,
whereupon they can check their voicemail as if they were using the internal
phone. I found no other way of doing this.
If you patch your app_voicemail.c, I have V1.44 from CVS as of
12/11/2003, with this diff file, and recompile the app_voicemail.so module
and install it in /usr/lib/asterisk/modules and then, from the command line
of Asterisk, do:
unload app_voicemail.so
load app_voicemail.so
you should have the new feature, all without having to stop and restart
asterisk.
Good luck, and let me know if it works for you.
-Brian
--- app_voicemail.c.fcs Thu Dec 11 12:55:25 2003
+++ app_voicemail.c Sat Feb 28 16:21:15 2004
@@ -1083,7 +1083,7 @@
char prefile[256]=;
char fmt[80];
char *context;
- char *ecodes = #;
+ char *ecodes = *#;
char *stringp;
time_t start;
time_t end;
@@ -1117,12 +1117,12 @@
if (mkdir(dir, 0700) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n, dir,
strerror(errno));
if (ast_exists_extension(chan, strlen(chan-macrocontext) ?
chan-macrocontext : chan-context, o, 1, chan-callerid))
- ecodes = #0;
+ ecodes = *#0;
/* Play the beginning intro if desired */
if (strlen(prefile)) {
if (ast_fileexists(prefile, NULL, NULL) 0) {
if (ast_streamfile(chan, prefile, chan-language)
-1)
- res = ast_waitstream(chan, #0);
+ res = ast_waitstream(chan, *#0);
} else {
ast_log(LOG_DEBUG, %s doesn't exist, doing what we
can\n, prefile);
res = invent_message(chan, vmu-context, ext, busy,
ecodes);
@@ -1138,6 +1138,10 @@
silent = 1;
res = 0;
}
+ if (res == '*') { /*break out to main vm*/
+ free_user(vmu);
+ return(100);
+ }
if (!res !silent) {
res = ast_streamfile(chan, INTRO, chan-language);
if (!res)
@@ -1156,6 +1160,10 @@
free_user(vmu);
return 0;
}
+ if (res == '*') { /*break out to main vm*/
+ free_user(vmu);
+ return(100);
+ }
if (res = 0) {
/* Unless we're *really* silent, try to send the beep */
res = ast_streamfile(chan, beep, chan-language);
@@ -2678,6 +2686,9 @@
}
res = leave_voicemail(chan, ext, silent, busy, unavail);
LOCAL_USER_REMOVE(u);
+ if (res == 100) { /*The user requested vm main*/
+ res = vm_execmain(chan, NULL);
+ }
return res;
}
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[Asterisk-Users] Re: res_motv: Request for comment
One thing that the BSD open source operating system projects do, and many other projects for that matter, which Asterisk does not seem to do, is put CVS ID tags in the source files of the package itself. If ID tags were put into the source files, and even embedded in strings so that theyshowed up in the binary files too, that would go a long way toward helping users determine which version of Asterisk they had, and where they were relative to the current state of the development tree. It seems like this change requires no real coding, just adding a line or two to each source file, and CVS does the rest for you by bumping the version numbers as changes come in. Another advantage of this approach, is that users can succinctly and accurately point out which versions of which modules work and which ones contain critical bugs. Then you can say things like: File res_moh.c, V1.25 and later contains the fix you're looking for. Just a thought. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Resetting Grandstream HT-286 to factory default settings?
Hello. I just purchased a Grandstream HT-286 from Chagres Technologies. When I initially set this up, I accidentally mistyped the new http password to get into the unit, and I cannot now log into the web server on the device. the user manual has this to say about how to reset the device to factory defaults, but I do not know how to enter a MAC address which contains letters -- do I pretend I'm dialing a name and use the numbers associated with the letters of the MAC address? When I try to do this, it doesn't reset, and tells me my numbers are invalid. Any suggestions on how to restore this box to factory freshness? -thanks -Brian [...] 99 RESET Enter 9 to confirm the RESET Enter MAC address to restore factory default setting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote retrieval of voicemail, a question
Hello. I'm running an asterisk system where the voicemail box numbers match the extensions to which they belong. The phone numbers from the PSTN which access the system are mapped to specific extensions, and if there's no answer, they forward to their respective mailboxes so callers can leave messages for the owners of the extensions. Without adding an additional voicemail only access number from the PSTN, I would like owners to be able to call their extensions and retrieve their messages through the PSTN. I've looked at the app_voicemail.c file in the Asterisk source tree, and I see how to do it with a source code change, i.e. allow the user to press * while the outgoing message is playing, and jump to voicemailmain and proceed to do generic voicemail authentication. However, I'm wondering if there's a way to do the same thing, that I've not thought of, which can be done without modifying the source code itself, i.e. through configuration changes in either voicemail.conf, extensions.conf, or through some other mechanism I've not thought of. I'm assuming here, that what I want is something others wanted before me, and that they've found a solution of which I'm not aware. Can anyone enlighten me? Many thanks in advance for any suggestions. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Gateway of choice is?
A cisco 1760 router, with a pair of dual FXO cards in it will work fine. We've been using a couple of these for years, and they're quite reliable, sound good, and behave themselves with Asterisk, using SIP. Not the cheapest, perhaps, but a good choice. If you want to save money, buy a used Cisco 2600 router, and use the same dual FXO cards, they're just as good. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk on FreeBSD 4.9?
I don't know if this helps, but I've been running our office IP phone system on Asterisk, on a NetBSD-1.6.1 system for over a month now, with no trouble at all. The functionality is limited at the moment, due to the lack of the features provided by the zaptel drivers, but I hope to remedy that in the not-too-distant future. -Brian Message: 10 Date: Tue, 13 Jan 2004 20:27:07 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk on FreeBSD 4.9? Reply-To: [EMAIL PROTECTED] On Tue, Jan 13, 2004 at 12:24:20PM -0500, Jason T. Nelson wrote: love to be able to use Asterisk under FreeBSD. I've browsed the archives and perceived what appears to be a slightly hostile attitude towards those who ask about Asterisk support of other free operating systems even without using Digium hardware. Is this Linux-specific bias intentional or accidental? I would call it historical. Asterisk was first developed on Linux, and little attention was paid to portability. This is changing, though there are still Linuxisms in the code. I would hesitate to consider it stable yet on anything other than Linux, but YMMV. I personally would like to see Asterisk portable to any *nix with pthreads, and am working to make this happen. As always help in the form of patches, testing or accounts for building and testing on less common types of systems are appreciated. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP
Hello. The Cisco 7905 and 7920 phones are basically the same phone, with the 7920 having a built-in ethernet switch. Sip and Skinny images are available for these phones on the Cisco web site if you hav a CCO account. I believe you select which image you want to run at boot time, with the OS7920.TXT file. (If you're familiar with the way this works with the Cisco 7940 and 7960 phones, you'll understand the procedure for getting these phones to boot the desired image.) Essentially, you put the version number of the image in the OS7920.TXT file, and use the S or M parameter in that file to determine whether you want an mgcp/skinny image or a Sip image. If you load a sip image into the phone, it should work quite wel with Asterisk. If you want to continue debugging and fixing the skinny code in Asterisk, then load the mgcp/skinny image into the phone. Note: the software can be found on the Cisco site under the Software Center link on the CCO registered user page. Hope that helps. -Brian Message: 5 Date: Tue, 13 Jan 2004 14:49:36 +0100 From: Jan Czmok [EMAIL PROTECTED] To: [EMAIL PROTECTED], [EMAIL PROTECTED] Subject: [Asterisk-Users] Again: 7920 Cisco IP Phone Skinny SIP Reply-To: [EMAIL PROTECTED] hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: -- Phone auth's with AP Phone gets IP from DHCP TFTP Server Phone loads OS7920.TXT Phone loads SEPmacaddr.CNF.XML Phone loads xmlDefault.conf.xml Phone registeres to Asterisk Phone gets registered Phone gets Info/Dial/Stuff from Asterisk Phone gets Line Info SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY LineStatReqMessage SKINNY LineStatMessage SKINNY SoftKeySetReqMessage SKINNY SoftKeySetResMessage SKINNY OffHookMessage SKINNY SetSpeakerModeMessage SKINNY OnHookMessage SKINNY DisplayPromptStatusMessage SKINNY DisplayPromptStatusMessage SKINNY DisplayPromptStatusMessage But if you look at the Support of the 7920 in Callmanager Express, you get a file named cmterm_7920.3.3-01-02-021.bin so i was investigating further. so i wrote cmterm_7920.3.3-01-02-021 in OS7920.TXT and suddenly the Cisco 7920 shows Upgrading Firmware :-) Unfortunately for some reason it did not accept the firmware, but it still tries to load it. Some additional info: - The 7920 is requesting cmterm_7920.3.3-01-02-021^J.bin (so with an Ctrl-J in it), so you have to rename the file. I also got the information from documents that the 7920 is running in 7960 emulation mode, so draw your own conclusions in regards of SIP possiblity :-) I tried to use some 7960 images, but did not succeed :-( Would appreciate some help in this issue :-) --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP
Sorry for my confusion. I'm thinking of the Cisco 7912, not the 7920. You're absolutely right. For the 7920, only mgcp/skinny is available from cisco. So, I guess it's debugging and hackery for Asterisk with respect to this phone. I plan to get my hands on one in the next month or so, and see if I can make it go with Asterisk's Skinny module. -Brian On Jan 13, 11:54pm, Jan Czmok wrote: } Subject: Re: [Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny SIP } Brian Buhrow ([EMAIL PROTECTED]) wrote: } } Hello. The Cisco 7905 and 7920 phones are basically the same phone, } } Hi Brian. 7905 is a normal desktop phone. 7920 is the WiFi Phone build } from cisco. } } } } [snip] } } with the 7920 having a built-in ethernet switch. Sip and Skinny images } are available for these phones on the Cisco web site if you hav a CCO } account. I believe you select which image you want to run at boot time, } with the OS7920.TXT file. (If you're familiar with the way this works with } the Cisco 7940 and 7960 phones, you'll understand the procedure for getting } these phones to boot the desired image.) Essentially, you put the version } number of the image in the OS7920.TXT file, and use the S or M parameter in } that file to determine whether you want an mgcp/skinny image or a Sip } image. } If you load a sip image into the phone, it should work quite wel with } Asterisk. } If you want to continue debugging and fixing the skinny code in } Asterisk, then load the mgcp/skinny image into the phone. } } [snip] } } see in my original mail, already tried this ... } } --jan } } } -- } Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. } Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] -- End of excerpt from Jan Czmok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sip phones on the same extension?
Hello. I think I understand your suggestion, but don't understand how that's any different than the one I came up with. What I want, is to be able to define a specific extension, and then have any external SIP phones register with that extension that want to. It's important that multiple phones be able to register with the same extension simultaneously. Then, I can define something like: exten = 300,1,Dial(SIP/300,15|t) and all phones registered to extension SIP/300 will ring. The number of phones existing on that extension at any given time is unknown, and Asterisk should be able to keep a list of all devices which are currently registered on a given extension, even if it has seen another device register to the same extension. To guard against number stealing, one could restrict the registration of a given phone number to a single password, but allow that password to be used as often and from where ever. So, for example, if my extension is 300, and my password is JustForFun, I should be able to program any number of SIP phones to register as extension 300, and as long as they know the magic password, JustForFun, Asterisk will permit all of them to register as SIP/300. Then, if someone calls 300, they'll all ring simultaneously, and which ever phone gets picked up first, gets the call. This doesn't appear to be how Asterisk works at the moment. Am I wrong about this? -Brian Message: 9 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip phones on the same extension? Date: Wed, 24 Dec 2003 13:24:53 -0600 Reply-To: [EMAIL PROTECTED] In sip.conf: [phone1] type=peer host=dynamic [phone2] type=peer host=dynamic [phone3] type=peer host=dynamic in extensions.conf: [default] exten = 0,1,Dial(SIP/phone1SIP/phone2SIP/phone3,30,T) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sip phones on the same extension?
Hello. I'm sorry I wasn't clear. In the original question I asked, I said that I found the same work around that was suggested on this list. Since the suggestion was there, and since I had posted my original work around in my original message, I thought there was something that I was missing with respect to the work around itself, and I was asking for clarification. The solution I have working at the moment, is exactly the one which was offered up. However, I don't like it, because it's a solution which doesn't scale. I was trying to assertain if Asterisk would do what I was envisioning, and which SER does very well, and if the fact that I couldn't think of a way was merely due to my lack of knowledge about Asterisk. It sounds like Asterisk doesn't work like this right now. Do folks think they'd find such a feature useful if I coded it up and sent it back to Digium? -thanks -Brian Message: 5 Date: Thu, 25 Dec 2003 13:20:51 -0600 (CST) From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Sip phones on the same extension? Reply-To: [EMAIL PROTECTED] Because the one you came up with isn't possible with asterisk at this time. On Thu, 25 Dec 2003, Brian Buhrow wrote: Hello. I think I understand your suggestion, but don't understand how that's any different than the one I came up with. What I want, is to be able to define a specific extension, and then have any external SIP phones register with that extension that want to. It's important that multiple phones be able to register with the same extension simultaneously. Then, I can define something like: exten = 300,1,Dial(SIP/300,15|t) and all phones registered to extension SIP/300 will ring. The number of phones existing on that extension at any given time is unknown, and Asterisk should be able to keep a list of all devices which are currently registered on a given extension, even if it has seen another device register to the same extension. To guard against number stealing, one could restrict the registration of a given phone number to a single password, but allow that password to be used as often and from where ever. So, for example, if my extension is 300, and my password is JustForFun, I should be able to program any number of SIP phones to register as extension 300, and as long as they know the magic password, JustForFun, Asterisk will permit all of them to register as SIP/300. Then, if someone calls 300, they'll all ring simultaneously, and which ever phone gets picked up first, gets the call. This doesn't appear to be how Asterisk works at the moment. Am I wrong about this? -Brian Message: 9 From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip phones on the same extension? Date: Wed, 24 Dec 2003 13:24:53 -0600 Reply-To: [EMAIL PROTECTED] In sip.conf: [phone1] type=peer host=dynamic [phone2] type=peer host=dynamic [phone3] type=peer host=dynamic in extensions.conf: [default] exten = 0,1,Dial(SIP/phone1SIP/phone2SIP/phone3,30,T) -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip phones on the same extension?
Hello. I'm a new Asterisk user, but I'm impressed with the flexibility and versatility of Asterisk, and am moving quickly to adopt it's main-line use in our company. Hopefully, you'll be hearing more from me as the project moves forward. Right now, though, I have a question about SIP peer registration. Right now, for our SIP-based phone,s, we're using the Sip Express Router product, which accepts sip registration requests and lets us route calls to any of the phones which register with SER. I am a semi-nomatic user, and can work at any of three different locations. Right now, my phones all sign up with SER, and register with the same telephone number. When someone dials that number, all three phones ring, and which ever one gets answered first, gets the call. When I tried to do this with Asterisk, sources from the cvs repository as of 12/18/2003, sip show peers only showed the most recent registration. This lead me to believe that if I dialed the number, only the most recently registered phone would ring. I was able to work around the problem by defining an umbrella extension which rings all three phones at the same time, but I'd like to have a way of dynamically adding phones to a given extension without having to necessarily rewrite the extensions.conf file, and I'd like calls from these extensions to show up from the master extension that folks should use to reach me. I imagine I could do something with pickup groups, but my understanding is that it is not true that all phones in a pickup group will necessarily ring just because they're a member of a given pickup group. The phones on this particular extension are many miles from each other, so one couldn't hear the other phone ring. Another work around is to put Asterisk behind SER, but this seems overly complicated, and I want to make sure that Asterisk doesn't do what I want before I pursue that path. Any suggestions on how to have multiple phones register with the same number in Asterisk? -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
