RE: [Asterisk-Users] Web based UA
By web based do you simply mean a UA that is *deployed* using the web (http) or do you also mean that to include tunneling of media over 80/443? Any Java based softphone could easily be turned into an applet, thus satisfying the web-based part of your query. An Active X component is nothing more than a resident program, and if you are looking at the ability to use this from any kiosk, then that kiosk would need to allow active X components, so you are kind of screwed. T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Moore Sent: Wednesday, February 25, 2004 1:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Web based UA I think xten is supposed to have an active X control version of their softphone that would probably do what you are talking about. On Wed, 25 Feb 2004, Michael Graves wrote: Hello All, Does anyone here have any experience with web based soft clients for *? I'm thinking about putting a page up on our corp web server that would let staff in the field connect to our in-house phone system via the internet. This could help staff making overseas calls while on trips, without demanding that they use a particular laptop/soft phone. They could use an PC on a broadband connection. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 -- Jonathan Moore Technology Coordinator Winfield Public Schools Office 316-221-5100 Fax 316-221-0508 ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] An example config for using a Wildcard X100P and a SIP phone?
Jason, Include your sip and extensions files so people can take a look. T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Monday, February 23, 2004 10:25 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone? Hello. I've just recently purchased the Asterisk Developers Kit so we can figure out how to get away from our Nortel system and go to IP based phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download). Either way, I can call the asterisk box and get their demo playing fine. I can even call the SIP phone I've hooked up when I call in from my cell phone to the asterisk box, and that works. I cannot call out with my SIP phone though. It'll dial, ring my cell phone twice and then give up and complain that its busy. Even if I try to answer the cell phone during the first ring. Does anyone have a config they could share with me on how to make this setup work? This sounds like it should be fairly trivial, but I've beaten my head against the wall on this for a few days. =) Thanks alot, 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] An example config for using a Wildcard X100P and a SIP phone?
Try moving your sip phone into its own context, instead of default (I use
sip) and create a [sip] section in your extensions.conf
Add a sepcific extension to test your outgoing, like :
exten = _5,1,Dial,Zap/1/800551212
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 1:02 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] An example config for using a Wildcard X100P
and a SIP phone?
Timothy,
I have minimally modified the demo files that came with Asterisk, so
what is posted below is most of the comments and the demo section
removed from the config files.
Thanks!
; SIP Configuration for Asterisk
;
[general]
port = 5060; Port to bind to
bindaddr = 0.0.0.0; Address to bind to
context = default; Default for incoming calls
[sipphone]
type=friend
username=sipphone
fromuser=Sipster; Specify user to put in from instead
of callerid
secret=password
host=dynamic
defaultip=192.168.1.201
amaflags=default; Choices are default, omit, billing,
documentation
accountcode=Sipster ; Users may be associated with an
accountcode tp ease billing
mailbox=431
--
extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
;CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1; Trunk interface
TRUNKMSD=1; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
[iaxtel700]
exten = _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[trunkint]
;
; International long distance through trunk
;
exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91800NXX,2,Congestion
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,2,Congestion
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,2,Congestion
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat = 9
include = local
include = trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat = 9
;include = default
;include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2}); Ring the interface, 20
seconds maximum
exten = s,2,Voicemail(u${ARG1}); If unavailable, send
to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1); If they press #,
return to start
exten = s,102,Voicemail(b${ARG1}); If busy, send to
voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If they press #,
return to start
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include = local
exten = 431,1,Dial,SIP/sipphone
Regovich, Timothy wrote:
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 10:25 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] An example config for using a Wildcard X100P and
a
SIP phone?
Hello.
I've just recently purchased the Asterisk Developers Kit so we can
figure out how to get away from our Nortel system and go to IP based
phones. I have a RH 9 box loaded with Asterisk (a very recent cvs
download).
Either way, I can call the asterisk box and get their demo playing fine.
I can even call the SIP phone I've hooked up when I call in from my cell
phone to the asterisk box, and that works.
I cannot call out with my SIP phone though. It'll dial, ring my cell
phone twice and then give up and complain that its busy. Even if I try
to answer the cell phone during the first ring.
Does anyone have a config they could share with me on how to make this
setup
RE: [Asterisk-Users] Codec Order / Preference
Really? Did you try disallow=all Allow=speex Allow=gsm Allow=alaw ? T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 23, 2004 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Codec Order / Preference You cannot specify the order of codec selection with Asterisk On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote: Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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 -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Room Monitor
Two coffee cans and a tight string? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamin W. Collins Sent: Wednesday, February 18, 2004 12:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Room Monitor On Tue, Feb 17, 2004 at 10:04:02PM -0800, David Liu wrote: Well use a Polycom IP 500 and put to auto answer and ringer off. Then you can use it as a room monitor device. Seems like that could do the trick. However, I was hoping for a sub $200 solution. Anyone know of a less expensive solution? -- Jamin W. Collins Remember, root always has a loaded gun. Don't run around with it unless you absolutely need it. -- Vineet Kumar ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Sip problem with IpDialog phone.
Turn sip debug on and forward the logs. A 481 means that a dialog was not correctly established. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista Sent: Thursday, February 12, 2004 6:28 PM To: Asterisk User List Subject: [Asterisk-Users] Sip problem with IpDialog phone. I have one of my IpDialog phones giving this error about once an hour. On the Asterisk server CLI I get this message. Got SIP response 481 Call Leg/Transaction Does Not Exist back from 204.241.XXX.XXX If I go to the phone and dial out it works and I no longer get the message. Also if I check the sip show channels I get 2 additional connections with unknown information for the IpDialog phone. Other then this message the phone work fine. But when the message comes up I can not dial call the phone. ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Can't connect KPhone to asterisk
Not ACK'ing an invite can be problematic for the statemachine. Without the ACK, the Dialog is not in acorrect state. As for the SDP goes, the KPHONE is offering what it can accept, and asterisk is doing the same. There is no restriction that they must match. You can change your offer in the ACK, or with a re-INVITE. As for the immediate transmission : yeah, it does seem a little strange doesn't it? But that is the way that I have seen almost all UAs work. The implication is that your offer must be a valid, not a conditional offer : when you say you accept GSM on port 8000, you better have a listener on 800 ready to go. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maciek Kaminski Sent: Wednesday, February 11, 2004 11:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't connect KPhone to asterisk Anyone managed to make KPhone work with Asterisk? For me it looks as if KPhone does not ACK transactions, i.e.: KPhone --INVITE-- Asterisk Asterisk --Trying -- KPhone Asterisk --OK -- KPhone KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone INVITES. Both timeouts. By the way: KPhone offers PCMU, GSM, iLBC in INVITE, Asterisk answers with PCMU and PCMA with doest not seem to be correct as it should answer with subset of codecs offered(as far as I understood SIP RFC). Another issue that bothers me is that Asterisk seems to start media transmission as soon as it send OK not after it received ACK. Begining of conversation may lost this way, isn't it? Asterisk and KPhone logs below: - Asterisk log: - Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;rport CSeq: 1974 INVITE To: sip:[EMAIL PROTECTED] Content-Type: application/sdp From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188 Call-ID: [EMAIL PROTECTED] Subject: sip:[EMAIL PROTECTED] Content-Length: 183 User-Agent: kphone/4.0 Contact: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp v=0 o=username 0 0 IN IP4 192.168.0.3 s=The Funky Flow c=IN IP4 192.168.0.3 t=0 0 m=audio 32778 RTP/AVP 0 97 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 11 headers, 9 lines Using latest request as basis request Sending to 192.168.0.3 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format GSM Found description format iLBC Capabilities: us - 12, them - 1030/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:4186 check_user: Setting NAT on RTP to 0 Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:5277 handle_request: Check for res for maciejka Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:1128 find_user: Call from user 'maciejka' is 1 out of 0 Looking for 700 in default Feb 11 17:12:36 DEBUG[81926]: chan_sip.c:3572 build_route: build_route: Contact hop: Maciek Kaminski sip:[EMAIL PROTECTED];transport=udp list_route: hop: sip:[EMAIL PROTECTED];transport=udp Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.3;rport From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188 To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0 Call-ID: [EMAIL PROTECTED] CSeq: 1974 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.3:5060 -- Executing Answer(SIP/maciejka-b4b6, ) in new stack We're at 192.168.0.2 port 15200 Answering with preferred capability 4 Answering with preferred capability 8 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.3;rport From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188 To: sip:[EMAIL PROTECTED];tag=as3b0a9ff0 Call-ID: [EMAIL PROTECTED] CSeq: 1974 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 153 v=0 o=root 3363 3363 IN IP4 192.168.0.2 s=session c=IN IP4 192.168.0.2 t=0 0 m=audio 15200 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 to 192.168.0.3:5060 -- Executing Festival(SIP/maciejka-b4b6, Press 1 to heaven press 2 to go to hell press 3 to disconnect.) in new stack == Parsing '/etc/asterisk/festival.conf': Found Feb 11 17:12:36 DEBUG[180236]: app_festival.c:318 festival_exec: Text passed to festival server : Press 1 to heaven press 2 to go to hell press 3 to disconnect. Feb 11 17:12:36 DEBUG[180236]: app_festival.c:395 festival_exec: Passing text to festival... Feb 11 17:12:36 DEBUG[180236]: app_festival.c:414 festival_exec: Passing data to channel... Feb 11 17:12:36 DEBUG[180236]: app_festival.c:424 festival_exec: Festival WV command Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;rport CSeq: 1974 INVITE To: sip:[EMAIL
RE: [Asterisk-Users] Can't connect KPhone to asterisk
Where is that quote from? Are rtpmaps marked as sendrecv or recvonly? There is nothing really that says that I couldn't receive mpeg audio, but only be able to send ulaw. If you don't want to start listening until you send the ACK, then don't send an SDP in the INVITE. Wait until the ACK to send it. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maciek Kaminski Sent: Wednesday, February 11, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can't connect KPhone to asterisk Regovich, Timothy wrote: Not ACK'ing an invite can be problematic for the statemachine. Without the ACK, the Dialog is not in acorrect state. As for the SDP goes, the KPHONE is offering what it can accept, and asterisk is doing the same. There is no restriction that they must match. You can change your offer in the ACK, or with a re-INVITE. Well, they must intersect: For streams marked as sendrecv in the answer, the m= line MUST contain at least one codec the answerer is willing to both send and receive, from amongst those listed in the offer. The stream MAY indicate additional media formats, not listed in the corresponding stream in the offer, that the answerer is willing to send or receive (of course, it will not be able to send them at this time, since it was not listed in the offer). As for the immediate transmission : yeah, it does seem a little strange doesn't it? But that is the way that I have seen almost all UAs work. The implication is that your offer must be a valid, not a conditional offer : when you say you accept GSM on port 8000, you better have a listener on 800 ready to go. Optimistic strategy... Maciek Kaminski ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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: [Fwd: [Asterisk-Users] Having problems with RTP packets and H old]
Why does the FXO gateway treat a lack of RTP packets as a dropped call (and
what heuristic does it use to determine?)
Until the SIP UA sends an actual BYE message, the Dialog should still be
considered active, regardless of the RTP that may or may not be happening.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clif Jones
Sent: Tuesday, February 10, 2004 1:33 PM
To: [EMAIL PROTECTED]; asterisk users
Subject: [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold]
If anyone is familiar with the SIP SDP handling routines I would appreciate
some
insight. The following problem that I found using Asterisk appears to be
improper
handling of a call put on hold when there is no music on hold:
[FXO gateway] [Asterisk]
[IP phone]
|---[INVITE s/SDP]|---[INVITE
s/SDP]|
| |
|
|[180 Ringing]|[180
Ringing]|
| |
|
|[183 Session Progress]---|---[200
OK/SDP]--|
| |
|
|[200
OK/SDP]-|[ACK]|
| |=== RTP
|
|[ACK]|
|
|=== RTP |
|
{IP phone puts caller on
hold}
| |-[INVITE/held
SDP]---|
| |
|
| |---[200
OK/SDP]--|
| |
|
|
|[ACK]|
| RTP (one-way)===|
|
| |
|
|--[BYE]--|
|
| |
|
|[200 OK]-|
|
When the IP phone puts the gateway on hold, Asterisk gets the re-INVITE with
held
media but Asterisk doesn't re-INVITE the gateway. The RTP traffic to the
gateway
stops so the gateway handles the condition as a lost connection. Shouldn't
asterisk
be forwarding the re-INVITE to the gateway unless MOH is enabled?
--
Notice: This e-mail message, together with any attachments, contains
information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New
Jersey, USA 08889), and/or its affiliates (which may be known outside the
United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as
Banyu) that may be confidential, proprietary copyrighted and/or legally
privileged. It is intended solely for the use of the individual or entity
named on this message. If you are not the intended recipient, and have
received this message in error, please notify us immediately by reply e-mail
and then delete it from your system.
--
___
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RE: [Fwd: [Asterisk-Users] Having problems with RTP packets and H old]
Can you send the sip debug messages along? That would help. I am
interested in what the original invites looked like dialog information) and
what the subsequent invite looks like.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clif Jones
Sent: Tuesday, February 10, 2004 2:16 PM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: Re: [Fwd: [Asterisk-Users] Having problems with RTP packets and H
old]
Good question. If you look at my original post, you will see that this
problem was discovered after
this feature was evidently added to our AudioCodes gateway GA
firmware. The beta code didn't
do this. They are probably trying to solve the problem of detecting
dropped calls from the IP side
but if this feature is not selectable you run into problems like
this. I'm actually beating them up over
this but I have not been impressed with their support as a company. I
have still failed to get DTMF
bridging via RFC2833 working 100%. If anyone has had success with
Audiocodes FXO SIP gateways
and Asterisk, I would like to know the magic formula that makes all this
work. :)
Regovich, Timothy wrote:
Why does the FXO gateway treat a lack of RTP packets as a dropped call (and
what heuristic does it use to determine?)
Until the SIP UA sends an actual BYE message, the Dialog should still be
considered active, regardless of the RTP that may or may not be happening.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clif Jones
Sent: Tuesday, February 10, 2004 1:33 PM
To: [EMAIL PROTECTED]; asterisk users
Subject: [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold]
If anyone is familiar with the SIP SDP handling routines I would appreciate
some
insight. The following problem that I found using Asterisk appears to be
improper
handling of a call put on hold when there is no music on hold:
[FXO gateway] [Asterisk]
[IP phone]
|---[INVITE s/SDP]|---[INVITE
s/SDP]|
| |
|
|[180 Ringing]|[180
Ringing]|
| |
|
|[183 Session Progress]---|---[200
OK/SDP]--|
| |
|
|[200
OK/SDP]-|[ACK]|
| |=== RTP
|
|[ACK]|
|
|=== RTP |
|
{IP phone puts caller on
hold}
| |-[INVITE/held
SDP]---|
| |
|
| |---[200
OK/SDP]--|
| |
|
|
|[ACK]|
| RTP (one-way)===|
|
| |
|
|--[BYE]--|
|
| |
|
|[200 OK]-|
|
When the IP phone puts the gateway on hold, Asterisk gets the re-INVITE
with
held
media but Asterisk doesn't re-INVITE the gateway. The RTP traffic to the
gateway
stops so the gateway handles the condition as a lost connection. Shouldn't
asterisk
be forwarding the re-INVITE to the gateway unless MOH is enabled?
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Banyu) that may be confidential, proprietary copyrighted and/or legally
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received this message in error, please notify us immediately by reply
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and then delete it from your system.
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RE: [Asterisk-Users] The Evil of type=friend explained, again ( wa s Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoin t IP 500)
Jeremy, There is one small flaw in your reasoning with the need to register. You said : You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device But this is not true if you want to maintain true presence information. If you do not register, no one who has subscribed to you will know that you are available. In many cases this is undesirable behavior. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Thursday, February 05, 2004 6:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500) David Liu wrote: Could you tell us a little bit how exactly it works? The wiki pages don't say much about type=friend, user, and peer. I tried using type=user but can't seem to register. A type=friend is simply both a type=user and type=peer using the same set of config directives. While a type=friend makes things almost trivial to get calls working in both directions, it will limit the flexibility of your config and even hinder some of the more advanced uses of Asterisk. For example: Say you want to use the same 'user' across many different Asterisk boxes, which of course will have different IP addresses. In this situation, you cannot have a host keyword in your Asterisk config stanza for the type=user, but the type=peer requires some host keyword. Thus, if you use a type=friend you will limit the use of that one username to whatever IP address is contained in the host keyword. You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device. To register you need to have a type=peer with a host=dynamic. Since in your type=friend config directive you had host=some.ip.address, while this may be this is fine to for the type=user, this same value also gets used for the type=peer, which makes it so you cannot register since the IP address is hard coded. So, either you do not need to register and things will Just Work(tm) or you will need to use separate type=user and type=peer config directives. I smell the beginnings of a Whitepaper here. Jeremy McNamara - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 2:47 AM Subject: Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500 mattf wrote: I have all of my Polycom's set to friend so I know that's not your problem. One day you too will get bitten by the type=friend's EVIL and you will see the light. Trust me, Jeremy McNamara ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Sip flow diagram?
Try RFC 3261 http://www.faqs.org/rfcs/rfc3261.html Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, February 04, 2004 12:45 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] Sip flow diagram? Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] SIP debug logs
Or you could modify the logger and have all SIP messages set at a different log level and have them go to a file (/var/log/messages/sip) for example. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geert Nijpels Sent: Tuesday, February 03, 2004 11:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP debug logs Steve Foy wrote: This strikes me as something that should be really very simple to do, but I can't figure it out. Is there a way of logging all SIP debuging info to a file somewhere? It would help me greatly! I dont know if it's possible using asterisk. You can use the command 'script -a filename' that will record everything at the prompt. 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] MeetMe Video option
Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the v flag on my extension for the meetme app? Thanks, Tim -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] MeetMe Video option
I have written my own. Java(JMF) based. It is pretty rudimentary, but does handle audio (gsm, ulaw) and video (jpeg and H263). Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Friday, January 30, 2004 1:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MeetMe Video option Regovich, Timothy wrote: Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. What video phone did you use? ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] MeetMe Video option
I was wondering if it was supported, and how. It seems to me that video conferencing is a different beast than audio conferencing because you cannot simply mix video like you can mix audio. The conferencing server would have to 1) mix the video by creating one aggregate outbound paneled type window, or 2) have each incoming stream sent to each registered listening stream, which is ok, as long as the client can handle multiple incoming streams reasonably (yes, I realize that this results in n*n bandwidth usage), or 3) the conference serer would need to designate a master video stream and ignore all other incoming streams. Each of these seem to be viable options, depending on what you want to do. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Friday, January 30, 2004 1:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MeetMe Video option Citeren Regovich, Timothy [EMAIL PROTECTED]: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Cool, what devices are you using ? Would love to try some :-) Is there something else that I need to be doing other than set the v flag on my extension for the meetme app? Hmm, don't think that's supported yet ?? -- Met vriendelijke groet, Florian Overkamp ObSimRef BV ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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: MeetMe Video option
So you are actually getting the video to come out though? I am not getting any outbound video RTP traffic at all. What settings do you have? If I get a chance this weekend I will take a look at the implementation and see what I can see. The mosaic thing should be pretty easy actually (really, just a scaling of each incoming stream and tiling them), but that won't work well for anything bigger than a 2x2 matrix, considering the bandwidth limitations of most users. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson Sent: Friday, January 30, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MeetMe Video option That's one of the things that's been on our (1control, I have nothing to do with Digium) wishlist/to do list that just hasn't gotten done yet. Currently, video in meetme is not supported. What we experience is the audio will conference with the other audio streams but the video just freezes. I was hoping to look into someday but I'm swamped with 1000 other things of higher priority. I have been thinking though, of some ways it could be supported, starting with the simplest and easiest: 1. First, if only 2 of the phones in the conference are video phones, allow them to exchange their video with each other, while having all of the audio streams conferenced as usual. 2a. The next step could be having each videophone rotate which stream it was showing for a few seconds (20 seconds maybe?). i.e. you could have 3 video calls mixed with several audio-only calls. Initially video call #1 would show #2's image, #2 would show #3's image, #3 would show #1's image for a few seconds, then rotate them by 1. Of course you don't need to show your own! :) Actually, ours has a picture-in-picutre in the corner so you can see yourself all the time anyway. 2b. The other option instead of time-rotating the images would be to try to show the image of whoever was talking. That kind of sounds like a pain to me, but maybe it's doable. 3. The really fancy thing would be to have Asterisk decode all of the video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them and send them to each client. That REALLY sounds like a pain to me, but again, maybe it's doable. Right now I'd be pretty happy with 2a though. - Matt Message: 3 From: Regovich, Timothy [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Date: Fri, 30 Jan 2004 13:07:46 -0500 Subject: [Asterisk-Users] MeetMe Video option Reply-To: [EMAIL PROTECTED] Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the v flag on my extension for the meetme app? Thanks, Tim ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] SIP: outbound calls
Hi all, Any advice on how to place a call from a SIP UA routed through *? Do I just place a sip call to [EMAIL PROTECTED]:5060 ? I am a little confused, since all of my Uas require registration for presence information. Thanks in advance, Tim -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Compiling problems with SuSE
Did anyone try compiling with optimizations off? I seemed to noticed that the default flag was an O9 or something. Try with -O1 or with -g ans see if it makes any difference. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Knuttgen Sent: Tuesday, January 20, 2004 9:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling problems with SuSE -Original Message- From: Uwe Klein [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Compiling problems with SuSE From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM We tried to use SuSE initially and had no luck compiling zaptel on either 8.2 or 9.0. We even had Digium take a look. After working on it for days we finally switched to Red Hat 9. Is there anyone who succeeded in compiling Asterisk with SuSE 8.2 or 9.0? HI Dustin, what kind of error did you get? something like this: pbx.c:581: warning: comparison between signed and unsigned pbx.c: In function `pbx_substitute_variables_temp': pbx.c:765: warning: comparison between signed and unsigned pbx.c:812: warning: comparison between signed and unsigned pbx.c: In function `pbx_builtin_hangup': pbx.c:4017: internal compiler error: Segmentation fault ?? I had problems with SuSE 8.2 and Asterisk from cvs dated ~12July2003 I got it fixed by adding 128MB of memory to the 32MB on this P200 machine. with 300MB of swap it should not have made a difference ( except taking forever ) but it did. G! UK -- Uwe Klein [mailto:[EMAIL PROTECTED] KLEIN MESSGERAETE Habertwedt 1 D-24376 Groedersby b. Kappeln, GERMANY phone: +49 4642 920 123 FAX: +49 4642 920 125 ___ 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 Uwe, I had a problem at the end when it does the depmod -a. We got an error with around ten modules. The only thing I could find related to the errors was something about PPP in the kernel or in the Makefile. Neither of which made any difference. Dustin ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] SMP kernel with X100P card
Hi all, Does anyone have any experience with running a X100P card with * in an SMP machine? I have plugged the card into a 4way 2.4 GHz server, and the hardware config seems ok -- the passthrough phone line works, the card has it's own IRQ on CPU0, and /proc/zaptel/1 doesn't show any errors. * seems to be ok as well, as I am able to register with my SIP proxy. (and I have set up the zaptel.conf and zapata.conf files for the card) However, I cannot see any channels for the card in *, and I get no answer when I ring the line. Can anyone offer anything that I can try? TIA Tim -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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
