Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
u probably should upgrade to 0.7.2 but as far as the caller id that would be from your sip.conf being set improperly add to your sip.conf callerid=Caller Name # for each sip entry and that should clear it up. On Sat, 2004-02-07 at 00:23, John Fraizer wrote: I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a sip only configuration currently. Everything is working except that caller ID is hosed. Say for example extension 100 calls extension 200. 200 sees 100 as the name but 200 as the number. IE, it gets its own number as the supposed CLID of the calling party. This is flat out wrong. Am I doing something wrong or is Asterisk just terribly broken with respect to sending caller ID information properly? Is this something that only effects Cisco phones? Thanks, John ___ 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 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] IAX Softphone Errors
I've been considering deploying an IAX softphone for some remote users that want to interface with my PBX. It seems as though IAXcomm just prints that it was rejected if they dial an extension unassigned on the PBX. Firefly on the other hand crashes if you dial an extension that doesn't atleast exist in the dialplan. I can't have that of course so I added a catch all group of extentions so if they dial any extension not defined prior it will just play invalid. I was wonder if anyone had any cleaner method to do this other than exten = _X,1,Macro(invalid). I wrote 12 variations to cover all the possible conditions I could think of but a program should crash over such an issue. I get a memory reading error with Firefly if I dial a # not defined in the context that my iax acc is part of. I noticed the Firefly network kept this from becoming an issue by making their dialplan give some feedback on all #'s dialed ___ 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] Asterisk under UML?
Scott Russ wrote: Does anyone know if/how well Asterisk will run under User Mode Linux? Will the ztdummy or zaprtc modules work with it? Thanks, Scott I have few boxes running it with no major problems. Ztdummy will not work becaause uml does not have real usb support. Zaprtc could work if you play with host kernel. (I have not had time to try yet). I have installed G729 into it as well. Works great. :) I can set you up, an UML for you to try if you wish. Ta SJ ___ 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] Interrupted musiconhold sound when silence suppression is enabled
Hi, On Fri, 6 Feb 2004, Andres wrote: * uses the incoming RTP Stream as a timing source for sending its outgoing Stream.. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppresion. I guess this explains why the music-on-hold goes totally silent (and fast-forwards) when I hit the microphone mute button on my 7960...? Could this be tied to zaptel timing instead (like meet-me)? Cheers, Vic Cross ___ 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] question for oh323 users
Anthony Law wrote: Hi Gus, Thanks for your reply. I have tried below and still didn't work. exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED] or exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] and now asterisk gives out below error Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel 'H323:8915' sent into invalid extension 's' in context 'default', but no invalid handler You must define a context for the incoming calls (section [register] in oh323.conf). You need someting like this: [register] context=demo gwprefix=1905 here is exactly what I have in extension.conf [general] static=yes writeprotect=no [default] include = demo [demo] exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] Any idea? Regards, Anthony Michael. ___ 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] Snom 100 Code Recommendation
G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. TIA, Jason ___ 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] OpenBSD 3.4 Patching
I'm trying to apply the latest OpenBSD patch off of bugs.digium.com and I'm having trouble. I get loads of errors (below) when I'm trying to apply this patch to a standard cvs checkout of * and suspect that this may be the root cause of my problems. Can anyone provide any advice as to what I need to do. TIA, Jason And the errors are... bash-2.05b# patch -C openbsd.patch Hmm... Looks like a unified diff to me... The text leading up to this was: -- |Index: Makefile |=== |+++ Makefile 5 Feb 2004 13:34:01 - -- Patching file Makefile using Plan A... Hunk #1 failed at 23. 1 out of 1 hunks failed--saving rejects to Makefile.rej Hmm... The next patch looks like a unified diff to me... The text leading up to this was: -- |Index: aesopt.h |=== |+++ aesopt.h 5 Feb 2004 13:34:02 - -- Patching file aesopt.h using Plan A... Hunk #1 failed at 148. 1 out of 1 hunks failed--saving rejects to aesopt.h.rej Hmm... The next patch looks like a unified diff to me... The text leading up to this was: -- |Index: channels/Makefile |=== |+++ channels/Makefile 5 Feb 2004 13:34:02 - -- Patching file channels/Makefile using Plan A... Hunk #1 failed at 52. Hunk #2 failed at 79. Hunk #3 failed at 129. Hunk #4 failed at 149. Hunk #5 failed at 170. 5 out of 5 hunks failed--saving rejects to channels/Makefile.rej Hmm... The next patch looks like a unified diff to me... The text leading up to this was: -- |Index: channels/chan_h323.c |=== |+++ channels/chan_h323.c 5 Feb 2004 13:34:03 - -- Patching file channels/chan_h323.c using Plan A... Hunk #1 succeeded at 26 with fuzz 2. Hunk #2 failed at 59. 1 out of 2 hunks failed--saving rejects to channels/chan_h323.c.rej Hmm... The next patch looks like a unified diff to me... The text leading up to this was: -- |Index: channels/h323/Makefile |=== |+++ channels/h323/Makefile 5 Feb 2004 13:34:04 - -- Patching file channels/h323/Makefile using Plan A... Hunk #1 failed at 1. Hunk #2 failed at 29. 2 out of 2 hunks failed--saving rejects to channels/h323/Makefile.rej Hmm... The next patch looks like a unified diff to me... The text leading up to this was: -- |Index: include/asterisk/frame.h |=== |--- include/asterisk/frame.h 4 Nov 2003 02:40:09 - 1.27 |+++ include/asterisk/frame.h 5 Feb 2004 13:34:04 - -- Patching file include/asterisk/frame.h using Plan A... Hunk #1 failed at 27. 1 out of 1 hunks failed--saving rejects to include/asterisk/frame.h.rej done ___ 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] s/asterisk mailinglists/asterisk forum/g ?
On Sat, 2004-02-07 at 09:06, Roy Sigurd Karlsbakk wrote: now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. I don't really think that making the mailing lists totally usable my moving them to a web based forum is the answer. Perhaps instead everyone that thinks the mailing list has too many messages/day to be useful should invest in a few moments to set up some basic filtering in their mail client. ___ 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] play_and_record: No audio available
Hello all, I have just installed the latest Asterisk csv on Slackware 9.1. I've configured the system using the example config in OnLamp's Asterisk: A Bare-Bones VoIP Example tutorial. [http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1] Using X-Lite SIP client on Mac OS X When trying to leave a voicemail on one of the extensions, the Asterisk console reports: Feb 7 23:26:17 WARNING[409618]: app_voicemail.c:1200 play_and_record: No audio available on SIP/2001-dc12?? and the resulting .wav file in /var/spool/asterisk is empty/invalid. I did find a thread (via google) that described someone with the same problem, who rectified it by re-installing. I've reinstalled, but to no avail. Can someone please point me in the right direction to troubleshoot this? Thanks Ryan ___ 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] s/asterisk mailinglists/asterisk forum/g ?
i search them just fine in Evolution. Filters to a different folder than my other mailing lists and works quite well. Different pop3 acc from my isp too =) Why use bandwidth on my colo'd boxes when I can use something I already paid for =) On Sat, 2004-02-07 at 10:30, Eric Wieling wrote: On Sat, 2004-02-07 at 09:06, Roy Sigurd Karlsbakk wrote: now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. I don't really think that making the mailing lists totally usable my moving them to a web based forum is the answer. Perhaps instead everyone that thinks the mailing list has too many messages/day to be useful should invest in a few moments to set up some basic filtering in their mail client. ___ 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 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] s/asterisk mailinglists/asterisk forum/g ?
It would be very helpful if the digest format could be improved, ie: ONE digest per day, and threads grouped within the digest. The format now is not very useful. I too am overwhelmed with 50-100 messages per day. Thanks! Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Suffill Sent: Saturday, February 07, 2004 4:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ? i search them just fine in Evolution. Filters to a different folder than my other mailing lists and works quite well. Different pop3 acc from my isp too =) Why use bandwidth on my colo'd boxes when I can use something I already paid for =) On Sat, 2004-02-07 at 10:30, Eric Wieling wrote: On Sat, 2004-02-07 at 09:06, Roy Sigurd Karlsbakk wrote: now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. I don't really think that making the mailing lists totally usable my moving them to a web based forum is the answer. Perhaps instead everyone that thinks the mailing list has too many messages/day to be useful should invest in a few moments to set up some basic filtering in their mail client. ___ 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 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 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] Caller-ID is being sent wrong. How to fix it?
Hi William,
Thanks for the reply. I don't understand why that information should have
to be set in sip.conf. It is already known. If I set the following on an
extension:
exten = 100,1,SetCallerID(${CALLERIDNUM})
I can see that the information is correct:
-- Executing SetCallerID(SIP/228-a94f, 228) in new stack
-- Executing Dial(SIP/228-a94f, SIP/100|20) in new stack
-- Called 100
But, the number when extension 100 rings, it gets 100 as the number
portion of the caller ID information.
John
William Suffill wrote:
u probably should upgrade to 0.7.2 but as far as the caller id that
would be from your sip.conf being set improperly add to your sip.conf
callerid=Caller Name # for each sip entry and that should clear it
up.
On Sat, 2004-02-07 at 00:23, John Fraizer wrote:
I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a sip only
configuration currently. Everything is working except that caller ID is hosed.
Say for example extension 100 calls extension 200. 200 sees 100 as the
name but 200 as the number. IE, it gets its own number as the supposed
CLID of the calling party.
This is flat out wrong. Am I doing something wrong or is Asterisk just
terribly broken with respect to sending caller ID information properly?
Is this something that only effects Cisco phones?
Thanks,
John
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Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. If you look at the sip debug of a call between extenstion 228 and extension 100, you can see what is causing the problem: -- Executing Dial(SIP/228-76d1, SIP/100|20) in new stack We're at 66.35.64.38 port 13694 Answering with preferred capability 4 Answering with preferred capability 8 Answering with preferred capability 256 Answering with non-codec capability 1 12 headers, 11 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK025b4e85 From: John Fraizer sip:[EMAIL PROTECTED];tag=as2e305230 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 07 Feb 2004 19:07:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 237 v=0 o=root 10168 10168 IN IP4 66.35.64.38 s=session c=IN IP4 66.35.64.38 t=0 0 m=audio 13694 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (NAT) to 24.33.239.118:5060 -- Called 100 It is this part that is causing it to get the wrong caller ID number: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK025b4e85 From: John Fraizer sip:[EMAIL PROTECTED];tag=as2e305230 To: sip:[EMAIL PROTECTED] Notice that we're inviting 100@ while claiming that the call is COMING from [EMAIL PROTECTED] It puts the right caller ID *name* in the invite but, the sip:100 in the from field is flat out wrong. It should be sip:228. Surely I am not the only one to notice that this is broken. John ___ 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] Snom 100 Code Recommendation
Jason Ross wrote: G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. I also have DTMF problems with Snom 200 running 2.03o, but haven't had the time or desire to dig too deep into it. I am running p2p with a sip gateway, so * is not in the picture and I have never changed code or reconfigured my gateway. I guess I have just been waiting for 2.03x release of the day to see if it gets better. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] s/asterisk mailinglists/asterisk forum/g ?
Good thing I am unemployed, so I have time to read this list. Every morning when I suck down my 200 emails from the list, I say to myself, I am going to implement some filters to help sort all these emails. But after blasting my way through the email, I am out of time and energy. Anyone have any filters they use on this list that may help me out. I have never set up any email filters. I run on a sun/sparc solaris 9 and use mozilla to read my email. A linux solution should be easy to get working on solaris. I know I should just learn how to do this myself, but I am too busy reading email. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. You are not setting the caller ID properly... callerid = string portion number portion If you want no string portion, then: callerid = number portion -- END OF LINE -MCP ___ 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] Snom 100 Code Recommendation
Jason Ross wrote: G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. TIA, Jason Do you have call progress turned on in zapata.conf ? If so, try turning it back off. I had tried turning it on and DTMF sent from my sip phones wasn't recognized anymore. -Lance ___ 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] Caller-ID is being sent wrong. How to fix it?
quote who=Robert Hajime Lanning quote who=John Fraizer OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. You are not setting the caller ID properly... callerid = string portion number portion If you want no string portion, then: callerid = number portion Also, it is the same syntax for the SetCallerID() application. The way you had it: callerid = 200 Sets the string portion to 200 and leaves the number portion null. The null number portion is what is causing you trouble. -- END OF LINE -MCP ___ 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] dial timeout not working
I've got a timeout set on a dial statement but I can't seem to get it to
work. NO matter what the value is set to, the phone just rings and
rings and rings
exten = _9NXX,1,Dial(${PSTN}/${EXTEN:1}|${timeout});dial
the number
exten = _9NXX,102,macro(dial-post-b,${EXTEN},${CFREF}) ;busy,
so go here
exten = _9NXX,2,macro(dial-post-u,${EXTEN},${CFREF})
;unavailable, so go here
I have set the timeout=15 and according to the console, the variable is
getting to the dial command ok.
-- Executing Dial(SIP/8006-5bcf, Zap/g1/7771000|15) in new stack
-- Called g1/7771000
-- Zap/1-1 answered SIP/8006-5bcf
-- Hungup 'Zap/1-1'
Anybody know what I'm screwing up ??
-Lance
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Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Robert Hajime Lanning wrote: quote who=John Fraizer OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. You are not setting the caller ID properly... callerid = string portion number portion If you want no string portion, then: callerid = number portion Um, yes I am setting the caller ID right. Asterisk isn't sending the invite message properly. [100] callerid= test name 1234 type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes The test name part gets sent but, like I said, if extension 100 calls extension 228, the phone at 228 sees the caller-ID as test name 228. This happens with Asterisk 0.5 and Asterisk 0.7.2 both. John ___ 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] All incoming Zap calls getting picked up as FAX calls!
There was a posting on this in December, but I didn't see this resolved - now I'm experiencing the same thing. I recently checked out asterisk using the 1.0 label as the asterisk.org now states. Perhaps the label is not kept current?? No matter, what I'm experiencing is every incoming call on a Zap channel is immediately assumed to be a fax. I checked the bugs database and didn't see anything substantial related to incoming faxes. 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] central voicemail with remote offices
I'm having trouble figuring this dialplan out. I have a central asterisk box that has voicemail and the auto attendant for incoming calls. Things get complex when I add three remote offices connected only through the Internet. The locals seem easy enough to handle. Redirecting through _2[0-2]N to another office makes sense. Where I'm having trouble is supporting a central voicemail system. If headoffice is where I want all the voicemail to reside, and an outside caller attempts a local. The voicemail statement is easy, its on the same server. But when someone from office-a calls someone in office-b, unless the call is routed through headoffice I can't see an easy way of redirecting for voicemail. Or if someone in office-a calls someone in the same office, how can I redirect the voicemail back to headoffice? And to further the madness, if my phone is in office-a, is there a way to know I have voicemail waiting for me at headoffice? I would like to do this because I expect the remote offices to have the occasional Internet connectivity troubles, but I don't expect our customers to care. They will still call the headoffice and attempt to contact local 1234. Perhaps I'm looking at this problem the wrong way. Has anyone been faced with a similar situation??? ___ 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] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer Um, yes I am setting the caller ID right. Asterisk isn't sending the invite message properly. [100] callerid= test name 1234 type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes The test name part gets sent but, like I said, if extension 100 calls extension 228, the phone at 228 sees the caller-ID as test name 228. This happens with Asterisk 0.5 and Asterisk 0.7.2 both. That is real interesting. It seems to work just fine for me. Though, I am running straigh out of CVS, but older than 0.7.2 release. My SIP phones (Grandstream) see CallerID just fine and my co-worker's SIP phones (Cisco 7960) work also. Can you send your extensions.conf? It has to be something in there. -- END OF LINE -MCP ___ 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] Caller-ID is being sent wrong. How to fix it?
I just set up an X-Lite client on extension and it shows the same behavior. Further information: When I call the X-Lite extension from my Cisco 7960 (extension 100), I get the following in the recent calls list: Name: Test SIPURL: [EMAIL PROTECTED] ProxyID: ENTERZONE The sip.conf entry for extension 100 is: [100] callerid=Test 100 type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes Asterisk is sending the name portion of the callerID properly but, the number portion is obviously wrong as you can see from the SIPURL that is saved by X-Lite in the recent received calls list. As I have been stating, it is sending the CALLED number and not the calling number. This is NOT the proper behavior and as a result, it is hosing caller ID. I could really not care any less about what name shows up. The sipurl that called has to be right though, otherwise callerid is worthless. It doesn't do any good to look at your missed calls list and have every one of them show YOUR phone number. If your asterisk server does not do this, please do me the favor of setting up two test extensions for me so I can try to figure out what is wrong here. You can lock me in a context where I can only call from one test extension to another. I just need to be able to verify what is going on so I can either get it corrected in my config (I don't think I have anything wrong) or get it acknowledged as a bug in Asterisk. Thanks, John John Fraizer wrote: Robert Hajime Lanning wrote: quote who=John Fraizer OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf. The behavior is the same. Caller-ID is sent as Name of Calling Party number of CALLED party instead of Name of Calling Party number of CALLING party like it should be. You are not setting the caller ID properly... callerid = string portion number portion If you want no string portion, then: callerid = number portion Um, yes I am setting the caller ID right. Asterisk isn't sending the invite message properly. [100] callerid= test name 1234 type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes The test name part gets sent but, like I said, if extension 100 calls extension 228, the phone at 228 sees the caller-ID as test name 228. This happens with Asterisk 0.5 and Asterisk 0.7.2 both. John ___ 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 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] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer If your asterisk server does not do this, please do me the favor of setting up two test extensions for me so I can try to figure out what is wrong here. You can lock me in a context where I can only call from one test extension to another. I just need to be able to verify what is going on so I can either get it corrected in my config (I don't think I have anything wrong) or get it acknowledged as a bug in Asterisk. I can do this, hold on. -- END OF LINE -MCP ___ 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] Caller-ID is being sent wrong. How to fix it?
Robert Hajime Lanning wrote:
That is real interesting. It seems to work just fine for me. Though, I am
running straigh out of CVS, but older than 0.7.2 release. My SIP phones
(Grandstream) see CallerID just fine and my co-worker's SIP phones (Cisco 7960)
work also.
Can you send your extensions.conf? It has to be something in there.
The portions that have anything to do with the two extensions I'm testing
with are:
[general]
static=yes
writeprotect=yes
[default]
exten = john,1,Dial(SIP/100,20)
[voicemail]
exten = 8500,1,Ringing
exten = 8500,2,Wait,1
exten = 8500,3,VoicemailMain2(${CALLERIDNUM})
exten = 8500,4,Hangup
exten = 8501,1,Ringing
exten = 8501,2,Wait,1
exten = 8501,3,VoicemailMain2
exten = 8501,4,Hangup
exten = 8502,1,Ringing
exten = 8502,2,Wait,1
exten = 8502,3,Voicemail2(u${CALLERIDNUM})
exten = 8502,4,Hangup
[extensions]
exten = 100,1,Dial(SIP/100,20)
exten = 100,2,Voicemail2(u100)
exten = 100,3,Hangup
exten = 100,102,Voicemail2(b100)
exten = ,1,Dial(SIP/,20)
exten = ,2,Hangup
[allaccess]
include = voicemail
include = extensions
Thanks,
John
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Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Would like to see a SIP debug * The invite from the caller phone to Asterisk * The invite from Asterisk to the called phone As well as the configs (extensions.conf and sip.conf) Can't reproduce in my servers. /O ___ 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] All incoming Zap calls getting picked up as FAX calls!
Look in the Asterisk Makefile and uncomment the line referring to OLD_DSP_ROUTINES. It's around line 119. On Sat, 2004-02-07 at 15:31, Darren Martz wrote: There was a posting on this in December, but I didn't see this resolved - now I'm experiencing the same thing. I recently checked out asterisk using the 1.0 label as the asterisk.org now states. Perhaps the label is not kept current?? No matter, what I'm experiencing is every incoming call on a Zap channel is immediately assumed to be a fax. I checked the bugs database and didn't see anything substantial related to incoming faxes. 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 -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] Caller-ID is being sent wrong. How to fix it?
Olle E. Johansson wrote: Would like to see a SIP debug * The invite from the caller phone to Asterisk * The invite from Asterisk to the called phone As well as the configs (extensions.conf and sip.conf) Can't reproduce in my servers. /O OK. Here is a call from extension 100 to extension 228. Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK149f400a From: John Fraizer 100 sip:[EMAIL PROTECTED];tag=000bbe40419b00532a4215e9-779f0059 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:46 GMT CSeq: 101 INVITE User-Agent: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 249 Accept: application/sdp v=0 o=Cisco-SIPUA 21234 22236 IN IP4 24.33.239.118 s=SIP Call c=IN IP4 24.33.239.118 t=0 0 m=audio 18846 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to 24.33.239.118 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK149f400a;received=24.33.239.118 From: John Fraizer 100 sip:[EMAIL PROTECTED];tag=000bbe40419b00532a4215e9-779f0059 To: sip:[EMAIL PROTECTED];tag=as0638308b Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=4844d22f Content-Length: 0 to 24.33.239.118:5060 Border2*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK3579a8a9 From: John Fraizer 100 sip:[EMAIL PROTECTED];tag=000bbe40419b00532a4215e9-779f0059 To: sip:[EMAIL PROTECTED];tag=as0638308b Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:46 GMT CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines Border2*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK3f142fb1 From: John Fraizer 100 sip:[EMAIL PROTECTED];tag=000bbe40419b00532a4215e9-779f0059 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:46 GMT CSeq: 102 INVITE User-Agent: CSCO/6 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=100,realm=asterisk,uri=sip:66.35.64.38,response=9d7ae43306bc23bb256068b8f4044017,nonce=4844d22f,algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 249 v=0 o=Cisco-SIPUA 21234 22236 IN IP4 24.33.239.118 s=SIP Call c=IN IP4 24.33.239.118 t=0 0 m=audio 18846 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to 24.33.239.118 : 5060 (NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 228 in allaccess list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK3f142fb1;received=24.33.239.118 From: John Fraizer 100 sip:[EMAIL PROTECTED];tag=000bbe40419b00532a4215e9-779f0059 To: sip:[EMAIL PROTECTED];tag=as4cba15e7 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 24.33.239.118:5060 *** THIS IS WHERE IT STARTS BREAKING *** -- Executing Dial(SIP/100-9284, SIP/228|20) in new stack We're at 66.35.64.38 port 10990 Answering with capability 2 Answering with capability 4 Answering with capability 8 Answering with non-codec capability 1 12 headers, 11 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4 From: John Fraizer 100 sip:[EMAIL PROTECTED];tag=as7e10d688 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 07 Feb 2004 22:57:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 234 v=0 o=root 12840 12840 IN IP4 66.35.64.38 s=session c=IN IP4 66.35.64.38 t=0 0 m=audio 10990 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101
Re: [Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled
Vic Cross wrote: Hi, On Fri, 6 Feb 2004, Andres wrote: * uses the incoming RTP Stream as a timing source for sending its outgoing Stream.. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppresion. I guess this explains why the music-on-hold goes totally silent (and fast-forwards) when I hit the microphone mute button on my 7960...? Could this be tied to zaptel timing instead (like meet-me)? Probably, but I don't think the developers are eager the code it. This has been discussed in great length before. Check the archives. Cheers, Vic Cross ___ 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 -- Andres Network Admin http://www.telesip.net ___ 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] All incoming Zap calls getting picked up asFAX calls!
I found the line, and uncommented it as you recommended. That worked - thanks!! As best I can tell, the problem still exists in the 1.0 stable code base. The good news is that I found a related bug 817 on the subject, although the fix has not made it to cvs yet. It's only known on this list and in the bugs database. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Saturday, February 07, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] All incoming Zap calls getting picked up asFAX calls! Look in the Asterisk Makefile and uncomment the line referring to OLD_DSP_ROUTINES. It's around line 119. On Sat, 2004-02-07 at 15:31, Darren Martz wrote: There was a posting on this in December, but I didn't see this resolved - now I'm experiencing the same thing. I recently checked out asterisk using the 1.0 label as the asterisk.org now states. Perhaps the label is not kept current?? No matter, what I'm experiencing is every incoming call on a Zap channel is immediately assumed to be a fax. I checked the bugs database and didn't see anything substantial related to incoming faxes. 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 -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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 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] Caller-ID is being sent wrong. How to fix it?
Robert Hajime Lanning wrote: I can do this, hold on. OK. I don't know what the deal is. Works fine on your server. Doesn't on mine. That is so strange. John ___ 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] Caller-ID is being sent wrong. How to fix it?
quote who=John Fraizer OK. I don't know what the deal is. Works fine on your server. Doesn't on mine. That is so strange. my version string is: CVS-01/31/04-04:24:34 Also, I noticed that your sip.conf entries are a bit different than mine. I am curious if canreinvite=no would change your situation. -- END OF LINE -MCP ___ 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] Caller-ID is being sent wrong. How to fix it?
On Sat, 7 Feb 2004, John Fraizer wrote: snip all the trace data Here are the configs: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 66.35.64.38 ; Address to bind to context = default ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls [100] type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes [228] type=friend username=228 secret=secret host=dynamic fromuser=228 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes [] type=friend username= secret=secret host=dynamic fromuser= context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes Remove fromuser= from your SIP statements. This overrides the caller-id data received with whatever is stated in fromuser -- Asterisk is doing exactly what you told it to. ;-) Hoo-roo, Vic Cross ___ 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] dial timeout not working
well onces Zap/1-1 answers SIP/8006 the timeout doesn't matter it thinks
its answered ( I bet this is an x100p isn' it)
bkw
On Sat, 7 Feb 2004, Lance Arbuckle wrote:
I've got a timeout set on a dial statement but I can't seem to get it to
work. NO matter what the value is set to, the phone just rings and
rings and rings
exten = _9NXX,1,Dial(${PSTN}/${EXTEN:1}|${timeout});dial
the number
exten = _9NXX,102,macro(dial-post-b,${EXTEN},${CFREF}) ;busy,
so go here
exten = _9NXX,2,macro(dial-post-u,${EXTEN},${CFREF})
;unavailable, so go here
I have set the timeout=15 and according to the console, the variable is
getting to the dial command ok.
-- Executing Dial(SIP/8006-5bcf, Zap/g1/7771000|15) in new stack
-- Called g1/7771000
-- Zap/1-1 answered SIP/8006-5bcf
-- Hungup 'Zap/1-1'
Anybody know what I'm screwing up ??
-Lance
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Re: [Asterisk-Users] IAX Softphone Errors
we'll be releasing a new version in the next few days that will fix that problem. Also, we've added the ability to store urls on the contact list. I'll post when we release the new version. - Original Message - From: William Suffill [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 07, 2004 7:50 PM Subject: [Asterisk-Users] IAX Softphone Errors I've been considering deploying an IAX softphone for some remote users that want to interface with my PBX. It seems as though IAXcomm just prints that it was rejected if they dial an extension unassigned on the PBX. Firefly on the other hand crashes if you dial an extension that doesn't atleast exist in the dialplan. I can't have that of course so I added a catch all group of extentions so if they dial any extension not defined prior it will just play invalid. I was wonder if anyone had any cleaner method to do this other than exten = _X,1,Macro(invalid). I wrote 12 variations to cover all the possible conditions I could think of but a program should crash over such an issue. I get a memory reading error with Firefly if I dial a # not defined in the context that my iax acc is part of. I noticed the Firefly network kept this from becoming an issue by making their dialplan give some feedback on all #'s dialed ___ 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 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] Snom 100 Code Recommendation
Jason Ross wrote: G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. I've been having busy problems with the 2.03x firmware versions, but no DTMF problems. I configured the phone for DTMF outband, with asterisk configured as dtmfmode=rfc2833. I'm running 2.02z. Check also: http://www.voip-info.org/wiki-SNOM+phones Kind regards, Geert ___ 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] X100P
Hi all, Just a quick question Does digium x100p support 3.3v pci (64 bit 66mhz) ? Regards, Soragan
[Asterisk-Users] Snom 200 MWI Button
I'm trying to get the MWI button to work with my Asterisk configuration. The snom is accepting and responding to the Message indications from *, but when I press the MWI button, it is dialing my extension (the one with the voice mail on it). I'm wondering if there is a way to specify what extension to dial to check email in the configuration, either the phone, or * itself. Asterisk Version 1/30/2003 checked out and compiled this evening Snom Version 2.03o (most recent auto-update) Any help would be greatly appreciated. At one point Mark had talked about adding a voicemail= directive in sip.conf on the mailing list at one point, however grepping the code doesn't reveal a feature like that at this time. Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Paul M. Oster
Re: [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?
Here is a suggestion for people interested in setting up a web based interface to the asterisk email list. Use fud forum : http://fud.prohost.org/ This is a forum system that can archive an email lists messages for easy searching and a nice web interface and it can be set up to allow posting of messages through the forum altough this may be undesirable. It is highly configurable. Fudforum if properly configured seems to offer a good way to improve access to an email list. I guess the question is whether the core users of this list want to improve access or whether they want to keep access limited to people who are at least committed enough to set up there email clients correctly which would be fair enough. Jamie -- James Pratt Phone no. : 03-33125787 Home Page : http://e-gakusei.org/ ___ 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] dial timeout not working
Brian West wrote:
well onces Zap/1-1 answers SIP/8006 the timeout doesn't matter it thinks
its answered ( I bet this is an x100p isn' it)
bkw
On Sat, 7 Feb 2004, Lance Arbuckle wrote:
I've got a timeout set on a dial statement but I can't seem to get it to
work. NO matter what the value is set to, the phone just rings and
rings and rings
exten = _9NXX,1,Dial(${PSTN}/${EXTEN:1}|${timeout});dial
the number
exten = _9NXX,102,macro(dial-post-b,${EXTEN},${CFREF}) ;busy,
so go here
exten = _9NXX,2,macro(dial-post-u,${EXTEN},${CFREF})
;unavailable, so go here
I have set the timeout=15 and according to the console, the variable is
getting to the dial command ok.
-- Executing Dial(SIP/8006-5bcf, Zap/g1/7771000|15) in new stack
-- Called g1/7771000
-- Zap/1-1 answered SIP/8006-5bcf
-- Hungup 'Zap/1-1'
Anybody know what I'm screwing up ??
-Lance
Yup, it certainly is...
So why is Zap/1-1 answering the call that it's making instead of
waiting for the far end to answer ?
-Lance
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Re: [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?
http://fud.prohost.org/ +1 *wow* Now that is a site for sore eyes! (yes, pun intended) -- I wonder how well the mailing list integration really does work. Wow. Regards, Andrew ___ 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] X100P
no why would it need to do that? If I put one in my 64 bit slots the machine won't boot. bkw On Sun, 8 Feb 2004, Soragan wrote: Hi all, Just a quick question Does digium x100p support 3.3v pci (64 bit 66mhz) ? Regards, Soragan ___ 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] Problems with ATA's locking up..
Anyone had any problems with ATA's running 3.0 software locking up? Thanks, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [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] modprobe wcfxs
Replying to my own e-mail, how tacky. Anyway, just for the record, I went out to the local computer shop looking for a new power supply, and ended up walking out $255 poorer, but with a new motherboard, 2.4Ghz CPU and 256M DDR memory, *and* a new case and power supply. 2 hours later, Debian was isntalled on an old disk, the kernel recompiled, zaptel recompiled, modprobes worked, and all is right with the world. So, solution *was* the motherboard, as the 2.2 PCI wasn't really supported until late 2000, it seems. Thanks for all the help, Tim On Fri, Feb 06, 2004 at 06:52:54PM -0500, Tim Sailer wrote: On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote: On Friday 06 February 2004 16:26, Tim Sailer wrote: OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) A common problem is forgetting to connect the 4-pin molex on the side Nope, plugged in. I even tested it to make sure the voltages were right. of the card. If you're still having problems, you could try a greater wattage power supply or a different motherboard. Hrm. A different mother board is out of the question right now. The power supply, maybe. ATX... It's strange, but while the TDM400P is up to the PCI spec, some motherboards are deficient. It is only when inserting a card which stresses the PCI spec to the max that you may wind up discovering this. This is an older motherboard, but it never had any problems driving things like video capture which tends to stress the bus. Also, if your TDM400P does not have a molex connector, you can get a free upgrade from the company that sold you the TDM400P. Brandy new from Digium. It had better be right. :) Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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 -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] ringing
When an extention (either ZAP or SIP) is dialed, the calling person hears just silence. Is it possible to get Ringing to work with this? It seems to cause people to hang up when there is silence. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] ringing
Tim Sailer wrote: When an extention (either ZAP or SIP) is dialed, the calling person hears just silence. Is it possible to get Ringing to work with this? It seems to cause people to hang up when there is silence. Tim http://www.voip-info.org/wiki-Asterisk+cmd+Dial -Lance ___ 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] ringing
On Sun, Feb 08, 2004 at 12:04:09AM -0500, Lance Arbuckle wrote: Tim Sailer wrote: When an extention (either ZAP or SIP) is dialed, the calling person hears just silence. Is it possible to get Ringing to work with this? It seems to cause people to hang up when there is silence. Tim http://www.voip-info.org/wiki-Asterisk+cmd+Dial Jeez. I'm tired. Thank you. I read that 3 times, and, only on the fourth time did I pick up on the 'r' option. What would be nice would be to have a different ring (like our big seimens), regular ring for an outside line, short-short for an inside line... Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] modprobe wcfxs
Yes, it does work on newer P4 motherboard. It does not work on Pentium II motherboard, tested with 3 brands, Abit, MSI, and Asus; all not working. But it does work with Pentium Pro motherboard (Intel original MB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Sunday, February 08, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] modprobe wcfxs Replying to my own e-mail, how tacky. Anyway, just for the record, I went out to the local computer shop looking for a new power supply, and ended up walking out $255 poorer, but with a new motherboard, 2.4Ghz CPU and 256M DDR memory, *and* a new case and power supply. 2 hours later, Debian was isntalled on an old disk, the kernel recompiled, zaptel recompiled, modprobes worked, and all is right with the world. So, solution *was* the motherboard, as the 2.2 PCI wasn't really supported until late 2000, it seems. Thanks for all the help, Tim On Fri, Feb 06, 2004 at 06:52:54PM -0500, Tim Sailer wrote: On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote: On Friday 06 February 2004 16:26, Tim Sailer wrote: OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) A common problem is forgetting to connect the 4-pin molex on the side Nope, plugged in. I even tested it to make sure the voltages were right. of the card. If you're still having problems, you could try a greater wattage power supply or a different motherboard. Hrm. A different mother board is out of the question right now. The power supply, maybe. ATX... It's strange, but while the TDM400P is up to the PCI spec, some motherboards are deficient. It is only when inserting a card which stresses the PCI spec to the max that you may wind up discovering this. This is an older motherboard, but it never had any problems driving things like video capture which tends to stress the bus. Also, if your TDM400P does not have a molex connector, you can get a free upgrade from the company that sold you the TDM400P. Brandy new from Digium. It had better be right. :) Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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 -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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 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] s/asterisk mailinglists/asterisk forum/g ?
hi all now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. regards roy ___ 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-Dev] s/asterisk mailinglists/asterisk forum/g ?
now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. This has been brought up several times and it's been shot down every single time. This time is no different. Web forums will not be viewed by the people who are actively developing asterisk since they're such a colossal waste of time, energy and bandwidth. It's so much harder to find info on a forum than it is on a mailing list -- hideous colour schemes, animated emoticons, l33t sp34k, broken PHP and database backends -- that's just the start. So, like most other web forums, they will inevitably stagnate into a breeding ground for newbies and fill with messages -- nay /posts/ -- about how asterisk sucks ass because nobody can get any help. These lists are searchable, that too has been brought up many times, and someone even wrote a search app that lets you do fuzzy matching and other neat tricks. I don't have the URL handy, but it should be in the archives for -users within the last 30 days. Regards, Andrew ___ 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] s/asterisk mailinglists/asterisk forum/g ?
Nope... works fine like it it! bkw On Sat, 7 Feb 2004, Roy Sigurd Karlsbakk wrote: hi all now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. regards roy ___ 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 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] s/asterisk mailinglists/asterisk forum/g ?
On Sat, 2004-02-07 at 10:06, Roy Sigurd Karlsbakk wrote: now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. While I myself am feeling a little overwhelmed by the mail traffic too, I don't like the idea of forums. Personally, I have yet to see a forum that I consider useful. All too often forums are slow, difficult to search and waste too much time throughout the day polling for new posts. I think a decent mail list archive with searching capabilities and sortable by date/thread/author should be fine. If the traffic is too much for you then unsubscribe and scan the archives from time to time. I'm involved with a few organizations that spent considerable time, effort and $$ setting up and managing forums only to find they are consistently under-utilized. I vote against forums. -joe -- Innovation Software Group, LLC - http://www.innovationsw.com Custom Internet and Computer Solutions Linux, UNIX, Java Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] s/asterisk mailinglists/asterisk forum/g ?
On 7 Feb 2004, Joe Phillips wrote: On Sat, 2004-02-07 at 10:06, Roy Sigurd Karlsbakk wrote: now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. While I myself am feeling a little overwhelmed by the mail traffic too, I don't like the idea of forums. Personally, I have yet to see a forum that I consider useful. All too often forums are slow, difficult to search and waste too much time throughout the day polling for new posts. I think a decent mail list archive with searching capabilities and sortable by date/thread/author should be fine. If the traffic is too much for you then unsubscribe and scan the archives from time to time. I'm involved with a few organizations that spent considerable time, effort and $$ setting up and managing forums only to find they are consistently under-utilized. I vote against forums. Ditto. I hate forums. They are annoying. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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
