Re: [Asterisk-Users] Issue with DTMF Tones - Codec Issues
I took a look at the NEAX brochures available from NEC's website. I may be wrong but I don't think you could change the way dtmf tones are sent from the PBX, but you should be able to send them out of band (with RTP, as per RFC 2833) from the cisco to the asterisk box. Generally, out of band dtmf is always better (when available) and more reliable than inband dtmf. Bear in mind that certain phones, such as grandstreams, do not work well with rfc2833 dtmf relaying, but work well with dtmf sent in SIP INFO messages. cheers On 8/16/05, Aaron W <[EMAIL PROTECTED]> wrote: > Thanks I give give that a try. One follow up question. If the call > is coming in via the PSTN, and going through the NEAX (PBX) then to > the Cisco, can I control the way the PBX sends the DTMF, or is the > cisco some how able to split out the DTMF tones from everything else? > > I was assuming that becuase I am going through the PBX, the cisco > would recieve the DTMF inband, and therefore it would have to send it > out also as inband. > > Thanks again > Aaron > > On 8/16/05, maka <[EMAIL PROTECTED]> wrote: > > just a suggestion, but why don't you try using RFC2833 dtmf relay > > between the cisco and the asterisk box. > > > > use dtmfmode=rfc2833 in sip.conf, and you can also set the dtmf mode > > per peer in sip.conf > > also, if you use inband dtmf, this would only work with u-law and > > a-law, and not g729. > > > > on the cisco, enter > > Router(config-dial-peer)# dtmf-relay rtp-nte > > in dial-peer configuration mode. > > > > I recently had problems with a cisco gw forwarding pstn dtmf digits to > > my asterisk box, and rfc2833(which is what rtp-nte stands for in > > cisco's terms) solved it successfully. > > > > > > cheers > > > > On 8/16/05, Aaron W <[EMAIL PROTECTED]> wrote: > > > Topology: > > > PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP > > > server > > > > > > When I make a call to a VoIP user from the PSTN, the call gets routed > > > through the PBX, and Cisco. Because of that the DTMF tones are passed > > > inband, which I can hear on the VoIP end of the call. However, I have > > > one extension on asterisk set up so that I can check voice mail when > > > away from my phone. When I call that number again via the PSTN, and I > > > am prompted to enter my extension number Asterisk never "hears" the > > > dtmf tones. I have done some digging around, and my guess is that the > > > issue relates to the codec being used messing up the tones. > > > > > > Am I on the right track? Is there a ideal way to handle this? what do > > > others do? > > > > > > I have posted my sip.conf below. > > > > > > Thanks, > > > Aaron > > > > > > [general] > > > port = 5060 ; Port to bind to > > > bindaddr = 0.0.0.0 ; Address to bind to > > > context = default ; Default for incoming calls (default > > > context has no routing for security purposes) > > > ;dtmfmode=rfc2833 > > > dtmfmode=inband > > > srvlookup = yes > > > disallow=all; Disallow all codecs > > > ;allow=g729 ; Codecs that we allow (in order of > > > preference) > > > allow=ulaw > > > ;allow=alaw > > > allow=g729 > > > ;allow=ulaw > > > ;allow=all > > > > > > > > > [3120] > > > callerid=Aaron Walsh <3120> > > > type=friend > > > host=dynamic > > > canreinvite=no > > > qualify=yes > > > nat=yes > > > setvar=LDPREFIX=199 > > > context=XXX > > > secret=X > > > [EMAIL PROTECTED] > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > > I'm sick and tired of being sick and tired... > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- I'm sick and tired of being sick and tired... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can not dial more then 23 calls
We are testing our Asterisk server prior to deployment. The server has a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and one PRI for local calls. We are using sipp from two different stations routing a test number out the LD lines and another test number out the PRI line. We can not get more then 23 total active calls to connect to the test numbers, the test numbers terminate to another PBX that we can monitor. We have dialed out using cell phones to this other PBX while the test is happening and it connects, meaning it has more then 23 active calls on it. If we place more then 23 calls then it seems to 'queue' the extra calls, though not all of the extra calls complete after we stop adding new calls. They seem to get stuck in a queue or lost. We will send 200 calls through the Asterisk server and all but about 20 do eventually complete. Those 20 or so are stuck as Asterisk thinks the channels are busy with the calls when in fact there are no 'real' calls on the server. We can send 30 calls through the LD or PRI and only 23 are actually connected at a time. We can send 30 calls to both LD and PRI at the same time and still only a mixture of 23 calls are actually active at one time. So our issue seems to be located in our Asterisk server. Is there a way to limit or throttle an Asterisk server so that it will not place more then 'x' calls? We need to be able to support 48 calls. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problems with eyebeam - video phone
Thank you for your answer.
I didn't register on the domain of the Eyebeam software, actually I don't
understand how to do that!
I bouught 5 eyebeam activation keys and I am trying with the first 2 of
them
On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263" codec,
no other.
If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the
two video phone speak without any problem (but without any video)
If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the
first video phone call the second, the second answer and immediately
the call ends.
If Ilook at /var/log/asterisk/full, I see:
Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi
completed, returning 0
Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial("SIP/551-eac0",
"SIP/552|25|tr") in new stack
Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL)
Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0
Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0
Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552
Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0
Aug 17 08:37:06 VERBOSE[14731]: -- Called 552
Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but
retaining packet) on '[EMAIL PROTECTED]'
Request 102: Found
Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing
Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102
Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Found
Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop:
Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0
Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2)
to SIP/552-ff46(524288)
Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make
SIP/551-eac0 compatible with SIP/552-ff46
Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse
counter
It seems the problem documented in bug
http://bugs.digium.com/bug_view_page.php?bug_id=0003709
but actually it is not exactly the same.
moreover: is there any way to put the patch described in
http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *)
in asterisk 1.0.9 and not asterisk CVS HEAD ?
Any help will be greatly appreciated.
Andrea
"Carlos Alperin"
<[EMAIL PROTECTED]
om.net>To
Sent by: "'Asterisk Users Mailing List -
asterisk-users-bo Non-Commercial Discussion'"
[EMAIL PROTECTED]
m.com cc
Subject
16/08/2005 20.48 RE: [Asterisk-Users] problems with
eyebeam - video phone
Please respond to
Asterisk Users
Mailing List -
Non-Commercial
Discussion
<[EMAIL PROTECTED]
ists.digium.com>
Hi,
I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I
only use H.263 and SIP. (G.729)
Now, the more important question is if you register on the domain on the
Eyebeam software. I found that this was the full secret about this.
Let me know your configuration on the Eyebeam side.
Regards,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, August 16, 2005 11:28 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] problems with eyebeam - video phone
I am trying to connect two Xten eyeBeam Video Phone
No problems in voice connecting.
I tryed to modify my sip.conf
[general]
language=it
videosupport=yes
; enable Asterisk video support
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=h263
allow=gsm
allow=ulaw
allow=alaw
; H.263 is our video codec
; al
Re: [Asterisk-Users] TE410P + SPANDSP fax problem
Ma Zhiyong schrieb:
...
Trace shows that the fax is received successfully.
Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX("Zap/94-1",
Hi,
sorry, I don't know the solution to your problem, but I would like
to know, how did you get that trace?
I'm looking for a reliable way to determine, whether TxFax did send
a fax completely. I also tried the option "debug", but never saw
such a trace.
Which version of spandsp are you using?
Roger.
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Re: [Asterisk-Users] Voicemail file permissions
On Tue, Aug 16, 2005 at 02:40:36PM -0400, hugolivude wrote: > I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). > > I'd like to give my Asterisk users the option of cleaning up their > voicemail mailbox from their Windows PCs. I set up Samba and added > all the users with restricted access to their mailbox only, but here's > the problem: > > The voicemail .wav files that Asterisk creates have root as both owner > and group. > Since the users do not have root privileges, they can't do > much with the files. BTW I'm not sure why the voicemail .wav files > have root as both owner and group because I followed the instructions > for running Asterisk other than root (see > http://www.voip-info.org/wiki-Asterisk+non-root). Which is a good thing regardless. > > Is there a way around this w/o giving everyone root privileges! Do you want to allow every user to delete another user's voicemail? If not, how do you sync voicemail users and samba users? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] X101P register map data please?
Hi Newbie, or would you prefer to be called VoIP(y)? :-) Thanks for the advice, It's great to hear from somebody that has suffered in the same way :-) I've cc'd in the dev and user lists mostly so that others looking for the same issue (FXO PCI Master Abort) can find some info! - hope you dont mind... On the card itself. I am assured by the vendor that they had the card up and running in a machine. Indeed, the vendor has taken one home, called me through it, and then given it to me... so I'm reasonably sure that under SOME situations, these cards work and I now have 2 of these cards! I'd be interested to know, are all the 3 cards you have had "identical" in terms of how they look? Do you still have them? Can we compare notes - (off list)? On the Zaptel driver... There are clearly inconsistencies in the driver, which I feel should be sorted out However, they are in code which people with working systems say is never reached. So.. yes, the driver should be cleaned up in order to handle the IRQ's better, but the question remains, why am I/you getting the Master Aborts in the first place... If the patch that I've done to the driver is the right thing to do, then maybe thats an answer for me/you/others. I still seem to have some problems, so I need to understand those first (see other post). At least the Master Abort doesn't bring the whole machine down. What I can't tell is why WE get the Master Aborts in the first place Speculation would be good! Any ideas? Cheers Mark. On 17 Aug 2005, at 07:16, VoIP Newbie wrote: Dear Mark, I got 3 X101P clone cards from 3 different vendors. One of them has the same problem like yours. Another one has echo issue. Only one from www.broad-tel.com works fine for me. You may want to contact the vendor and get one for yourself instead of modifying ZAPTEL software. Newbie On 8/16/05, Mark Burton <[EMAIL PROTECTED]> wrote: Hi, I've been trying to debug the problem with the X101P giving FXO PCI Master Aborts... I'm doing this blind, and I really need some info on the X101P's register map - or best of all, the conditions under which it can generate an IRQ with a mask of 0x10. I have so far set up the mask for the IRQ's in the interrupt handler (so the poor thing doesn't keep getting them)[as per previous post], then patched ztcfg so it actually starts the watchdog (which is assumed by the driver, but in reality doesn't happen - of course it doesn't need to, because under normal conditions there is no need for the watchdog - I guess?) That much gives me a system which runs, hits a PCI Master abort (or at least an IRQ with a mask of 0x10), and then stops the dma, masks the IRQ... then the watchdog starts the dma again, unmasks the IRQ, at which point it gets another IRQ before the next watchdog beat so the watchdog can't help. I have tried being a bit more brutal with the activities in the watchdog routine I just caused myself some kernel panic's :-) Again, any help appreciated! Cheers Mark. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
On 17 Aug 2005, at 02:26, Don Fanning wrote:
I've surmized that it's Voipbuster having issues. Paid up another euro
on the second account and it works fine. When their support gets
better, I'll have them work on the other account.
I've had similar "flakyness" with Voipbuster. Sometimes the call goes
through a dream, next time I either get "no authority found" or invalid
extension/context. For me it's 50/50
This seems odd.. I put it down to their "free" service ...
[Though, what s worse, If Voipbuster fails, then voipjet fails too,
in the same way, and that I REALLY dont understand! But I haven't got
on that case to Voipjet yet - so i dont know what the problem is...]
Cheers
Mark.
-Don
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Fanning
Sent: Tuesday, August 16, 2005 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
I added in a second account that does not have the 1 Euro deposit and
it
goes through.
What would make things so different?
(this time the number is to the NIST Atomic Clock)
---
*CLI> iax2 debug
IAX2 Debugging Enabled
-- Executing SetCallerID("SIP/100-d2c1", ""jfalcon"") in new stack
-- Executing Dial("SIP/100-d2c1",
"IAX2/[EMAIL PROTECTED]/0013034997111") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569]
VERSION : 2
CALLED NUMBER : 0013034997111
CALLING NAME: jfalcon
LANGUAGE: en
USERNAME: jfalcon
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185631973
-- Called [EMAIL PROTECTED]/0013034997111
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
AUTHMETHODS : 3
CHALLENGE : 188826810
USERNAME: jfalcon
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
FORMAT : 2
-- Call accepted by 213.61.187.146 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
ACK
Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRQ
Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
LAGRP
Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
-- Hungup 'IAX2/voipbuster/10'
== No one is available to answer at this time
-- Executing NoOp("SIP/100-d2c1", "DIALSTATUS=NOANSWER") in new
stack
-- Executing NoOp("SIP/100-d2c1", "HANGUPCAUSE=0") in new stack
-- Executing Dial("SIP/100-d2c1",
"IAX2/[EMAIL PROTECTED]/0013034997111") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00018ms SCall: 5 DCall: 0 [213.61.187.147:4569]
VERSION : 2
CALLED NUMBER : 0013034997111
CALLING NAME: jfalcon
LANGUAGE: en
USERNAME: jfalcontwo
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185631979
-- Called [EMAIL PROTECTED]/001303499
RE: [Asterisk-Users] IAX compatible phones
For example TEK SIP-IAX 323. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dr. Marios Moutzouris Sent: Wednesday, August 17, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAX compatible phones Hello, I would like to know which phones are IAX compatible. Thank-you Marios Moutzouris -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] X101P register map data please?
Hi Newbie, or would you prefer to be called VoIP(y)? :-) Thanks for the advice, It's great to hear from somebody that has suffered in the same way :-) I've cc'd in the dev and user lists mostly so that others looking for the same issue (FXO PCI Master Abort) can find some info! - hope you dont mind... On the card itself. I am assured by the vendor that they had the card up and running in a machine. Indeed, the vendor has taken one home, called me through it, and then given it to me... so I'm reasonably sure that under SOME situations, these cards work and I now have 2 of these cards! I'd be interested to know, are all the 3 cards you have had "identical" in terms of how they look? Do you still have them? Can we compare notes - (off list)? On the Zaptel driver... There are clearly inconsistencies in the driver, which I feel should be sorted out However, they are in code which people with working systems say is never reached. So.. yes, the driver should be cleaned up in order to handle the IRQ's better, but the question remains, why am I/you getting the Master Aborts in the first place... If the patch that I've done to the driver is the right thing to do, then maybe thats an answer for me/you/others. I still seem to have some problems, so I need to understand those first (see other post). At least the Master Abort doesn't bring the whole machine down. What I can't tell is why WE get the Master Aborts in the first place Speculation would be good! Any ideas? Cheers Mark. On 17 Aug 2005, at 07:16, VoIP Newbie wrote: Dear Mark, I got 3 X101P clone cards from 3 different vendors. One of them has the same problem like yours. Another one has echo issue. Only one from www.broad-tel.com works fine for me. You may want to contact the vendor and get one for yourself instead of modifying ZAPTEL software. Newbie On 8/16/05, Mark Burton <[EMAIL PROTECTED]> wrote: Hi, I've been trying to debug the problem with the X101P giving FXO PCI Master Aborts... I'm doing this blind, and I really need some info on the X101P's register map - or best of all, the conditions under which it can generate an IRQ with a mask of 0x10. I have so far set up the mask for the IRQ's in the interrupt handler (so the poor thing doesn't keep getting them)[as per previous post], then patched ztcfg so it actually starts the watchdog (which is assumed by the driver, but in reality doesn't happen - of course it doesn't need to, because under normal conditions there is no need for the watchdog - I guess?) That much gives me a system which runs, hits a PCI Master abort (or at least an IRQ with a mask of 0x10), and then stops the dma, masks the IRQ... then the watchdog starts the dma again, unmasks the IRQ, at which point it gets another IRQ before the next watchdog beat so the watchdog can't help. I have tried being a bit more brutal with the activities in the watchdog routine I just caused myself some kernel panic's :-) Again, any help appreciated! Cheers Mark. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX compatible phones
Hello, I would like to know which phones are IAX compatible. Thank-you Marios Moutzouris -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC astcc-config.conf card length question
The only thing it will do is always ask users for 10 digit lengths (I think). This may confuse your users but that is the worst that will happen. Darren Wiebe [EMAIL PROTECTED] Nate Kapi wrote: I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE REPLY, are you using an X101P
I bought 3 from 3 different vendors. One of them has echo issue. Another one has an issue regarding PCI master abort. Only one really works fine for me. These 3 cards use AMBIENT chip but with different layouts and SLICs. On 8/4/05, Mark Burton <[EMAIL PROTECTED]> wrote: > X101P with Ambient md3200 chip on it, with the zaptel wcfxo driver > Just an indication of how many people have got this to work would be > useful. > > Cheers > > Mark. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: florz patch for bristuff breaks compile on x86_64?
On Wed, 17 Aug 2005, Remco Barende wrote: After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Anyone managed to get bristuff with florz working on x86_64 arch? Thanks! Sorry for replying to my own message, I forgot to include the error: rm -f zaphfc.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 install -D -m 644 zaphfc.ko /lib/modules/`uname -r`/misc/zaphfc.ko install: cannot stat `zaphfc.ko': No such file or directory make: *** [installlinux26] Error 1 hfc-pci driver installed. Press to continue, or + to abort. All other packages from bristuff compile fine after florz, just not zaphfc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] florz patch for bristuff breaks compile on x86_64?
After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an Athlon64) I also wanted to get the latest bristuff. Unfortunately bristuff without florz causes the box to kernel panic within hours (console will complain about bad frame received something). It seems however that the florz patch will not work for x86_64 arch. Bristuff -0.2.0-RC8j compiles fine without the florz patch, but after applying the patch zaphfc will not compile anymore (the patch applies cleanly). Anyone managed to get bristuff with florz working on x86_64 arch? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 8 FXS in Asterisk Server
Get a 8-port FXS gateway from www.broad-tel.com. That is the single box you need. On 8/16/05, Roland Zagler <[EMAIL PROTECTED]> wrote: > Hello everyone, > > I want to build an Asterisk Box where i need 8 FXS interfaces > to connect 8 phones to. The problem is, that there is only one > PCI slot available. What i have is 4 USBs 2.0 interfaces free > (if this helps). > > So here's my question: how am i going to do this? > > i tried to find any PCI cards supporting 8 FXS interfaces, but > without success. does anyone know such hardware? > > Thanks in advance, > Roland > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC astcc-config.conf card length question
I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Called Party Identification on Polycom IP501
Try quotes and no spaces between name and number. Callerid="first last"<2471> > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Anthony Rodgers > Sent: Tuesday, August 16, 2005 5:31 PM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Re: Called Party Identification on Polycom IP501 > > Hi Damon, > > It's not working SIP to SIP - I am wondering if there is something I am > missing in my * config. > > What I see on the Polycom display is: > > To:2471 > 2471 > > Called party entry in sip.conf (calling party entry is identical): > > [2471] > type=friend > context=internal > callerid=C* M <2471> > secret= > host=dynamic > nat=no > canreinvite=no > dtmfmode=rfc2833 > [EMAIL PROTECTED] > > The called party entry in phone2471.cfg (calling party entry is > identical): > > > > > > reg.1.label="2471" reg.1.type="private" reg.1.auth.userId="2471" > reg.1.auth.password=""/> > > msg.mwi.1.callBack="*98"/> > > > > Am I missing anything? > > Regards, > > Anthony > > > That is very dependent on how the call egresses from *, ISDN, POTS, > > SIP, > > ??? > > Who are you calling? > > > > > > If I recall correctly it will work when you call another extension on > > the * box, but the signaling for that info does not exists in > > PRI/T1/POTS, so it is not an * issue if you area calling out, * cant > > get > > the info from the telco, so * cant send it to the phone. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P + SPANDSP fax problem
Time and time again. CHECK YOUR Span clock src./bOn Aug 16, 2005, at 10:18 PM, Ma Zhiyong wrote: Hi, I just setup a fax server by spandsp. But it doesn't look good. Because each fax I received from my fax machine is not completed. I use te410p work with it. While the voice call is good. Any ideas? Trace shows that the fax is received successfully. Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX("Zap/94-1", "/var/spool/asterisk/FAX/1124251267.284.tif") in new stackAug 17 12:01:46 DEBUG[19571]: ==Aug 17 12:01:46 DEBUG[19571]: Pages transferred: 1Aug 17 12:01:46 DEBUG[19571]: Image size: 1728 x 355Aug 17 12:01:46 DEBUG[19571]: Image resolution 7700 x 3850Aug 17 12:01:46 DEBUG[19571]: Transfer Rate: 9600Aug 17 12:01:46 DEBUG[19571]: Bad rows 66Aug 17 12:01:46 DEBUG[19571]: Longest bad row run 22Aug 17 12:01:46 DEBUG[19571]: Compression type 2Aug 17 12:01:46 DEBUG[19571]: Image size (bytes) 0Aug 17 12:01:46 DEBUG[19571]: ==Aug 17 12:01:49 DEBUG[19571]: ==Aug 17 12:01:49 DEBUG[19571]: Fax successfully received.Aug 17 12:01:49 DEBUG[19571]: Remote station id: xxAug 17 12:01:49 DEBUG[19571]: Local station id: Aug 17 12:01:49 DEBUG[19571]: Pages transferred: 1Aug 17 12:01:49 DEBUG[19571]: Image resolution: 7700 x 3850Aug 17 12:01:49 DEBUG[19571]: Transfer Rate: 9600Aug 17 12:01:49 DEBUG[19571]: ==Aug 17 12:01:51 VERBOSE[2999]: -- Channel 0/1, span 4 got hangup___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
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[Asterisk-Users] TE410P + SPANDSP fax problem
Hi,
I just setup a fax server by spandsp. But
it doesn't look good. Because each fax I received from my fax machine is
not completed.
I use te410p work with it. While the voice
call is good.
Any ideas?
Trace shows that the fax is received
successfully.
Aug 17 12:01:10 VERBOSE[19571]: --
Executing RxFAX("Zap/94-1", "/var/spool/asterisk/FAX/1124251267.284.tif") in
new stackAug 17 12:01:46 DEBUG[19571]:
==Aug
17 12:01:46 DEBUG[19571]: Pages transferred: 1Aug 17 12:01:46
DEBUG[19571]: Image size: 1728 x
355Aug 17 12:01:46 DEBUG[19571]: Image resolution 7700 x
3850Aug 17 12:01:46 DEBUG[19571]: Transfer
Rate: 9600Aug 17 12:01:46 DEBUG[19571]: Bad
rows 66Aug
17 12:01:46 DEBUG[19571]: Longest bad row run 22Aug 17 12:01:46
DEBUG[19571]: Compression type 2Aug 17 12:01:46
DEBUG[19571]: Image size (bytes) 0Aug 17 12:01:46 DEBUG[19571]:
==Aug
17 12:01:49 DEBUG[19571]:
==Aug
17 12:01:49 DEBUG[19571]: Fax successfully received.Aug 17 12:01:49
DEBUG[19571]: Remote station id: xxAug 17 12:01:49 DEBUG[19571]:
Local station id: Aug 17 12:01:49 DEBUG[19571]: Pages transferred:
1Aug 17 12:01:49 DEBUG[19571]: Image resolution: 7700 x 3850Aug 17
12:01:49 DEBUG[19571]: Transfer Rate: 9600Aug 17
12:01:49 DEBUG[19571]:
==Aug
17 12:01:51 VERBOSE[2999]: -- Channel 0/1, span 4 got
hangup
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[Asterisk-Users] Solved: Unable to load module for TE406P
It works out that name "Unified t4xxp/t2xxp driver" is not accepted anymore by 2.6.13 kernel. Need to remove "/" for it to load properly > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Boris Bakchiev > Sent: Monday, 15 August 2005 18:17 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Unable to load module for TE406P > > Hi, > > I'm unable to load wct4xxp module for TE406P card. > > I've compiled 2.6.13-rc6, got latest CVS (as of today) of zaptel, but > when I try to load the module I get this: > > kobject_register failed for Unified t4xxp/t2xxp driver (-13) > [kobject_register+53/73] kobject_register+0x35/0x49 > [bus_add_driver+62/153] bus_add_driver+0x3e/0x99 > [driver_register+55/58] driver_register+0x37/0x3a > [pci_register_driver+120/134] pci_register_driver+0x78/0x86 > [pg0+945848335/1069265920] t4_init+0xf/0x22 [wct4xxp] > [sys_init_module+199/462] sys_init_module+0xc7/0x1ce > [syscall_call+7/11] syscall_call+0x7/0xb > > Can anyone point me into right direction to solve this? > Thanks! > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: New Beta IAX Statistics Program
Scott Bussinger wrote: >>Hi, we have put together a small application for Windows to allow you to >>check IAX network statistics. > > > The application seems to work fine, but could you point to any information > that helps interpret the information that it displays? I've seen the little > bit that the new jitterbuffer documentation provides, but I'd like a bit > more indepth information on what to do with the raw information. > > Thanks! Be seeing you. Ok. Local vs Remote are pretty self explanatory. Return Trip Time: Time taken for the return trip (ping) Jitter: Say you have 70ms ping and it sometimes goes up to 100ms, then this is 30ms jitter. Loss %: Percentage of packets which are lost Loss Count: Total number of packets which have been lost in this test Packets: Total number of packets Delay: The local/remote delay - this one I'm not too sure on Dropped: The number of packets that were dropped (presumably to decrease the size of the jitter buffer) Out of Order: Number of packets which have arrived out of sequence, i.e. 1,2,3,5,4,6,7,8 - 4 is out of order. Let me know if you have any other questions. :) I'm also looking at developing an IAX Client version that works by reading commands from a text file and printing responses to another text file. I have this pretty much working now for a proprietary customer but wonder if it would be useful to some other developers (~400K exe + ~250K dll). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 way calling
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I have three way calling on my Bell lines, so b4 Asterisk, 3 way calls were established by establishing call 1, pressing the Flash key, dialing the other party, and finally pressing the Flash again to join everyone. The three way call only used one line. I'm able to do three way calling with Asterisk using the Flash key as well, but I just discovered that when I do it w/ Asterisk (press the Flash key, dial the other party, press Flash again) BOTH of my POTS lines (I only have 2) get used up. Any ideas on what's wrong w/ my configuration? Essentially, I need to be able to send the "Flash signal" to Bell in order to establish the three way call on a single FXO line, but I also need Asterisk to act on the "Flash signal" in order to transfer calls. Any ideas? Thanks, Hugh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax
Steve Underwood wrote: That message isn't really well thought out, Sorry, I'll do better next time. :-) Lee. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk support T1 E&M Wink/Wink voice channels on any Digium/Sangoma hardware?
Yes. It does work. I had it working with a Varion card into an S8700 system with a TN464 DS1 circuit pack sitting in a IP 600 cabinet earlier this year as a proof of concept. Double check that your *ANI*DNIS* settings in the Definity setup match what you're expecting on the Asterisk side of things and vice versa. On 8/16/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hi, > > Did anyone manage to connect either Digium or Sangoma T1 card to any other > PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya > Definity to an Asterisk box with T100P and so far no luck. (I know I can do > so with ISDN PRI, but need an additional ISDN processor card for Definity.) > I tried to connect Definity to Cisco 3640 CCME (call manager express) to > test the link and all settings and it connects just fine with E&M Wink/Wink. > I just wanted to know that it is possible... is it? So far, I found only > this as a guide: > > http://www.asteriskguru.com/tutorials/em.html > > Thanks a lot, Dmitry. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All Page ??
Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute script on Answer
Roland Zagler wrote: Hello, i was wondering if it is possible to execute an AGI or shell script when a channel is answered. Does anyone have suggestions on how to do this? Look at the M option: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 501 Firmware
Hi, I have sort of a strange problem. We just bought a few polycom 501 phones from our supplier, but they shipped them with the SIP firmware, though we need them in H323. Does anyone here know where one can get ahold of the H323 firmware. I tried going through Polycom’s site directly and can’t seem to find anything. Please let me know Thanks S. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Execute script on Answer
Hello, i was wondering if it is possible to execute an AGI or shell script when a channel is answered. Does anyone have suggestions on how to do this? Thanks in advance, Roland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does Asterisk support T1 E&M Wink/Wink voice channels on any Digium/Sangoma hardware?
Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link and all settings and it connects just fine with E&M Wink/Wink. I just wanted to know that it is possible... is it? So far, I found only this as a guide: http://www.asteriskguru.com/tutorials/em.html Thanks a lot, Dmitry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TxFax -> RxFax on same machine hangs
Steve Underwood schrieb: ... If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will Hi, the setup is: TxFax (Box A) Dial(Zap...) (Box A, Digium Card) v PSTN v Box B, Digium Card Dial(IAX2...) (Box B) v RxFax (Box A) TxFax and RxFax ran on Box A. The PSTN call was accepted at Box B and then forwarded via IAX2 to Box A. RxFax and TxFax did nothing, and were never terminated, and thus needed an expicit "Hangup" command. Regards, Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Result from TxFax
Hi, there are some messages indicating, that TxFax is able to return -1 on failure. Well, I tried a lot but didn't succeed. I even sent a fax to a phone set, picked up the hand set and waited until timeout of TxFax. There is no difference to success. The only thing I could determine, is, when the other party hangs up. Any other case called the next priority in the dialplan. Is there any reliable mean, to check, whether a fax is really sent successfully and complete? Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP "agent" phone w/ headset
I would go with an ATA like the SPA-1001 and an analog set that mets your requirements. That's more than likely going to be your best bet. On 8/16/05, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 16:01, Tue 16 Aug 05, Colin Stefani wrote: > > I have a call center where we're looking at converting it from a > > traditional PBX w/ digital phone "agent" sets (keyless phones) that have > > headsets to a SIP based environment. > > > > I am having trouble finding anything on the market that resembles this > > in the VoIP world. > > > > For reference, we're currently using Inter-Tel Agent Sets, which are > > basically a digital phone with out any keypad, buttons or handset, just > > a line input and a headset jack. I need the equivalent. > > > > I know the first thing you think is why don't you use the agent's PC as > > the VoIP client and do a softphone, however I need to protect the caller > > from getting cut off should the PC crash/die/etc. While paranoid it's > > something where a regular endpoint like an ATA or SIP phone would be the > > best option. > > SIP phones and ATA's can die too. > * can die too > heck even your power can go down (hurricane, terrorist > attack, etc, etc) > > A properly configured pc with a softfone can be as stable as > a normal phone, it all depends what the users are doing with > it (I have had bad experience with pc's where users can > install their own stuff etc). > I have a workstation with an uptime of over 500 days. This > email was written on it. > > The problem will be the 'without keyped, buttons or > handset'. I'm not aware of a SIP device that has only a line > button and a headset and nothing else. > Judging on the setup you outlined, the agents are not able > to transfer the call to admin/other_user/parking_slot. They > are only able to receive calls, and that's all. > > If so, you can create them as 'user' only in sip.conf > That way they are only able to receive calls, but not make > calls. The interface to * is something you choose. > Of course phones/ATA's are less error-sensitive as pc's, > cause you can configure them. Just make sure noone can guess > the username/password for the ATA/phone config interface. > > Hope this helps, > -- > Michiel van Baak > http://michiel.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Why is it drug addicts and computer afficionados are both called users?" > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 way calling?
On Tue, 16 Aug 2005 18:05:58 -0400, hugolivude wrote: >Here's the problem - I only have 2 FXO lines. I can call one person, >transfer them to the conference room then hang up. Now 1 FXO is used. > I can then call another person and transfer them to the conference >room and hangup. Now the BOTH lines are used. I can then dial the >local extension for the conference room and I have a three way call >but that's as far as I get. > >I've tried establishing a three way call on a single line, but then I >cannot seem to transfer the call to the conference room. > >There's another problem as well. I have three way calling on my Bell >lines, so b4 Asterisk, 3 way calls were established by pressing the >Flash key and the three way call only used one line. I just >discovered that when I do the same thing w/ Asterisk (press the Flash >key, dial the other party, press Flash again) BOTH of my POTS lines >get used up. Any ideas on what's wrong w/ my configuration? > I'd not bother with using the flash based 3 way calling. Instead I'd setup an account with an ITSP and make the outbound calls via IP, preferabbly via IAX2. That way to can reach out to as many people as your bandwidth allows. Simply. Conveniently. Add one IP based DID and you can let others call in to your conference via IP. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
I've surmized that it's Voipbuster having issues. Paid up another euro
on the second account and it works fine. When their support gets
better, I'll have them work on the other account.
-Don
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Fanning
Sent: Tuesday, August 16, 2005 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
I added in a second account that does not have the 1 Euro deposit and it
goes through.
What would make things so different?
(this time the number is to the NIST Atomic Clock)
---
*CLI> iax2 debug
IAX2 Debugging Enabled
-- Executing SetCallerID("SIP/100-d2c1", ""jfalcon"") in new stack
-- Executing Dial("SIP/100-d2c1",
"IAX2/[EMAIL PROTECTED]/0013034997111") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569]
VERSION : 2
CALLED NUMBER : 0013034997111
CALLING NAME: jfalcon
LANGUAGE: en
USERNAME: jfalcon
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185631973
-- Called [EMAIL PROTECTED]/0013034997111
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
AUTHMETHODS : 3
CHALLENGE : 188826810
USERNAME: jfalcon
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
FORMAT : 2
-- Call accepted by 213.61.187.146 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
ACK
Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRQ
Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
LAGRP
Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
-- Hungup 'IAX2/voipbuster/10'
== No one is available to answer at this time
-- Executing NoOp("SIP/100-d2c1", "DIALSTATUS=NOANSWER") in new
stack
-- Executing NoOp("SIP/100-d2c1", "HANGUPCAUSE=0") in new stack
-- Executing Dial("SIP/100-d2c1",
"IAX2/[EMAIL PROTECTED]/0013034997111") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00018ms SCall: 5 DCall: 0 [213.61.187.147:4569]
VERSION : 2
CALLED NUMBER : 0013034997111
CALLING NAME: jfalcon
LANGUAGE: en
USERNAME: jfalcontwo
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185631979
-- Called [EMAIL PROTECTED]/0013034997111
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00018ms SCall: 00148 DCall: 5 [213.61.187.147:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00052ms SCall: 00148 DCall: 5 [213.61.187.147:4569]
AUTHMETHODS : 3
CHALLENGE : 529526436
USERNAME: jfalcontwo
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00227ms SCall: 5 DCall: 00148 [213.61.187
Re: [Asterisk-Users] TxFax -> RxFax on same machine hangs
Roger Schreiter wrote: Hi, I noticed a strange behaviour: Faxing using spandsp (TxFax) from my asterisk box to my old, common fax machine at home works fine. Faxing from the same box to my office pc-fax (Hylafax) also worke fine. Receveiving faxes on my asterisk box using spandsp (RxFax) also works fine. It is a PSTN number connected to the digium card of that asterisk box. Then I faxed from my asterisk box (TxFax) to that PSTN number, my asterisk box answers the call, TxFax and RxFax, both start, but no tif file is created in the directory, where normally the fax files are created. Using show channel I can still see both apps, TxFax and RxFax, even after half an hour. Then I stopped using soft hangup, and tried again several times: Same result. So, do I have to avoid to fax to myself in any case, because I risk to produce those never terminating jobs, which probably consume some resources? How can I enable asterisk to fax to itsself? Well, it won't be the normal operation, but when allowing clients to fax, it can happen by chance, that someone faxes to another user on the same machine without knowing it. Thanks for any hints! If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax
Lee Howard wrote: Brian West wrote: No it is really about faxing. As someone that has first hand knowledge of the case outlined on Groklaw, it is in fact about faxing. Go read the two patents very carefully! If you email it you break 638, if you store it you break 021. How, then, do these patents themselves not violate Brooktrout's own portfolio of earlier fax-specific patents covering virtually the same things? You could similarly ask how Brooktrout got patents on doing things common at the time. Or why people are still wary of using GIF, even though the Unisys patent on LZW has expired but IBMs identical patent has not. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 dialing problem
Sounds like you have a DTMF mode problem. Check that you are using RFC2833 for dtmf signaling. I had the same thing happen with my dialing of *98 to check voicemail..It would transpose it in to 9*8, as if the * was being some sort of a tab key. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: "Craig Bruenderman" <[EMAIL PROTECTED]> To: Sent: Tuesday, August 16, 2005 11:55 AM Subject: [Asterisk-Users] Polycom 501 dialing problem When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax
Hi Lee, Lee Howard wrote: Brian West wrote: Just an FYI http://www.groklaw.net/article.php?story=2005080914234645 Although Groklaw seems to think that these suits are about faxing, I don't think that they really are. See: http://www.hylafax.org/archive/2005-08/msg00107.html That message isn't really well thought out, and seems more wishful thinking than analysis. You are correct that what Asterisk does is not relevant to the patents, as it is too new. The rest of what you said doesn't make a lot of sense. Be assured, the people being sued are in the FAX business. This is very much about FAX. I used to do a lot of store and forward voice mail and FAX through e-mail and proprietary communications in the mid 90's. However these days there is really little need to do this with voice. Store and forward now is pretty much only about FAX. DID type FAX and voice to a retrieval box, and FAX and voice to e-mail has been used heavily for a long time. Also FAX to e-mail to FAX, very much like T.37 is old. I did that in 1994, and I didn't think it was very novel at the time. One of those patents has its origin in 1988, but Brian West has found www.pan.com proudly claim they did FAX to e-mail and e-mail to FAX gatewaying in 1987. :-) With enough searching I think they patents can be shot down thoroughly. However, you can never count on that, even if the evidence is clear and overwhealming. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail file permissions
My suggestion would be, use the externnotify=/usr/bin/myapp feature in voicemail.conf to chown the permissions to something else. Since they are root, asterisk should have no problem deleting and moving them around with less privileges. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: "hugolivude" <[EMAIL PROTECTED]> To: Sent: Tuesday, August 16, 2005 11:40 AM Subject: [Asterisk-Users] Voicemail file permissions I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I'd like to give my Asterisk users the option of cleaning up their voicemail mailbox from their Windows PCs. I set up Samba and added all the users with restricted access to their mailbox only, but here's the problem: The voicemail .wav files that Asterisk creates have root as both owner and group. Since the users do not have root privileges, they can't do much with the files. BTW I'm not sure why the voicemail .wav files have root as both owner and group because I followed the instructions for running Asterisk other than root (see http://www.voip-info.org/wiki-Asterisk+non-root). Is there a way around this w/o giving everyone root privileges! Thanks, Hugh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring from cell phone
Its left as default, and when I press the # nothing happens, but the remote
caller doesn't hear the DTMF tone.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, August 16, 2005 10:05 AM
Subject: Re: [Asterisk-Users] Transferring from cell phone
On 22:31, Mon 15 Aug 05, Chris Coulthurst wrote:
I set up a context to allow me to call in to my * server (via Teliax in
this case using IAX2) from my cellphone, and let me do a number of
things, including dial other extensions, AND dial outbound again so
callers could see my proper work CallerID when I use this service. Is
there a way to be able to transfer calls to other extensions of my
asterisk server FROM the cell phone/ This isn't a Zap channel, so I'm a
bit lost, but did specify the 'T' option in dial. Here's my context. Is
this possible to do??
[aa_chris_disa]
exten => s,1,Read(DIALNUM,custom/enter-num-then-pound,21)
exten => s,2,Playback(connecting)
exten => s,3,GotoIf($[${LEN(${DIALNUM})} < 5 ]?4:8) ; IF SHORTED THAN 5,
its internal so dial internal
exten => s,4,SetCallerID("Chris Mobile" <205>)
exten => s,5,Dial(Local/[EMAIL PROTECTED]/n) ;DIAL INTERNAL EXTENSION
exten => s,6,Playback(call-terminated)
exten => s,7,Goto(aa_chris_start,s,1)
exten => s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS
DIALED, ITS LOCAL, PREPEND THE AREA CODE
exten => s,9,SetCIDNum(99)
exten => s,10,Dial(${IPTRUNK}/360${DIALNUM},,T)
exten => s,11,Dial(SIP/[EMAIL PROTECTED],,T)
exten => s,12,Playback(all-circuits-busy-now)
exten => s,13,Goto(aa_chris_start,s,1)
exten => s,14,SetCIDNum(99) ;NUMBER ISNT 7, OR LESS THAN 5 SO
AREA CODE WAS ADDED
exten => s,15,Dial(${IPTRUNK}/${DIALNUM},,T)
exten => s,16,Dial(SIP/[EMAIL PROTECTED],,T)
exten => s,17,Playback(all-circuits-busy-now)
exten => s,18,Goto(aa_chris_start,s,1)
exten => i,1,Goto(aa_chris_start,s,1)
Did you modify features.conf ?
If not, what happens if you puch the # key on your cellphone
when connected to * ?
If you did change it, try the key you configured there for
transfer :)
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
"Why is it drug addicts and computer afficionados are both called users?"
___
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RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
I added in a second account that does not have the 1 Euro deposit and it
goes through.
What would make things so different?
(this time the number is to the NIST Atomic Clock)
---
*CLI> iax2 debug
IAX2 Debugging Enabled
-- Executing SetCallerID("SIP/100-d2c1", ""jfalcon"") in new stack
-- Executing Dial("SIP/100-d2c1",
"IAX2/[EMAIL PROTECTED]/0013034997111") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00013ms SCall: 00010 DCall: 0 [213.61.187.146:4569]
VERSION : 2
CALLED NUMBER : 0013034997111
CALLING NAME: jfalcon
LANGUAGE: en
USERNAME: jfalcon
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185631973
-- Called [EMAIL PROTECTED]/0013034997111
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00013ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 4ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
AUTHMETHODS : 3
CHALLENGE : 188826810
USERNAME: jfalcon
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00186ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
MD5 RESULT : 95fd16ba91a429b62028fc1ec6aa9cb5
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00186ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00188ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
FORMAT : 2
-- Call accepted by 213.61.187.146 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00188ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10014ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
ACK
Timestamp: 10014ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRQ
Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
LAGRP
Timestamp: 10002ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10002ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10729ms SCall: 00691 DCall: 00010 [213.61.187.146:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10729ms SCall: 00010 DCall: 00691 [213.61.187.146:4569]
-- Hungup 'IAX2/voipbuster/10'
== No one is available to answer at this time
-- Executing NoOp("SIP/100-d2c1", "DIALSTATUS=NOANSWER") in new
stack
-- Executing NoOp("SIP/100-d2c1", "HANGUPCAUSE=0") in new stack
-- Executing Dial("SIP/100-d2c1",
"IAX2/[EMAIL PROTECTED]/0013034997111") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00018ms SCall: 5 DCall: 0 [213.61.187.147:4569]
VERSION : 2
CALLED NUMBER : 0013034997111
CALLING NAME: jfalcon
LANGUAGE: en
USERNAME: jfalcontwo
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185631979
-- Called [EMAIL PROTECTED]/0013034997111
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00018ms SCall: 00148 DCall: 5 [213.61.187.147:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00052ms SCall: 00148 DCall: 5 [213.61.187.147:4569]
AUTHMETHODS : 3
CHALLENGE : 529526436
USERNAME: jfalcontwo
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00227ms SCall: 5 DCall: 00148 [213.61.187.147:4569]
MD5 RESULT : f53ebdd653ec18f40678a19b2eeece60
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00227ms SCall: 00148 DCall: 5 [213.61.187.147:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00222ms SCall: 00148 DCall: 5 [213.61.187.147:4569]
FORMAT : 2
-- Call accepted by 213.61.187.147 (format gsm)
-- Format for call is gsm
Tx-Fra
[Asterisk-Users] Re: Called Party Identification on Polycom IP501
Hi Damon, It's not working SIP to SIP - I am wondering if there is something I am missing in my * config. What I see on the Polycom display is: To:2471 2471 Called party entry in sip.conf (calling party entry is identical): [2471] type=friend context=internal callerid=C* M <2471> secret= host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 [EMAIL PROTECTED] The called party entry in phone2471.cfg (calling party entry is identical): reg.1.label="2471" reg.1.type="private" reg.1.auth.userId="2471" reg.1.auth.password=""/> msg.mwi.1.callBack="*98"/> Am I missing anything? Regards, Anthony That is very dependent on how the call egresses from *, ISDN, POTS, SIP, ??? Who are you calling? If I recall correctly it will work when you call another extension on the * box, but the signaling for that info does not exists in PRI/T1/POTS, so it is not an * issue if you area calling out, * cant get the info from the telco, so * cant send it to the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP "agent" phone w/ headset
On 16:01, Tue 16 Aug 05, Colin Stefani wrote: > I have a call center where we're looking at converting it from a > traditional PBX w/ digital phone "agent" sets (keyless phones) that have > headsets to a SIP based environment. > > I am having trouble finding anything on the market that resembles this > in the VoIP world. > > For reference, we're currently using Inter-Tel Agent Sets, which are > basically a digital phone with out any keypad, buttons or handset, just > a line input and a headset jack. I need the equivalent. > > I know the first thing you think is why don't you use the agent's PC as > the VoIP client and do a softphone, however I need to protect the caller > from getting cut off should the PC crash/die/etc. While paranoid it's > something where a regular endpoint like an ATA or SIP phone would be the > best option. SIP phones and ATA's can die too. * can die too heck even your power can go down (hurricane, terrorist attack, etc, etc) A properly configured pc with a softfone can be as stable as a normal phone, it all depends what the users are doing with it (I have had bad experience with pc's where users can install their own stuff etc). I have a workstation with an uptime of over 500 days. This email was written on it. The problem will be the 'without keyped, buttons or handset'. I'm not aware of a SIP device that has only a line button and a headset and nothing else. Judging on the setup you outlined, the agents are not able to transfer the call to admin/other_user/parking_slot. They are only able to receive calls, and that's all. If so, you can create them as 'user' only in sip.conf That way they are only able to receive calls, but not make calls. The interface to * is something you choose. Of course phones/ATA's are less error-sensitive as pc's, cause you can configure them. Just make sure noone can guess the username/password for the ATA/phone config interface. Hope this helps, -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called Party Identification on Polycom IP501
Anthony Rodgers wrote: Does anyone know if * can provide the "network signaling" required? If so, how? Not yet, no. I will be working on that after the 1.2 release of Asterisk is made, and we will be anxious for testers to try it out :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Done
---
*CLI>
*CLI>
*CLI>
-- Executing SetCallerID("SIP/100-1ba9", ""x"") in new stack
-- Executing Dial("SIP/100-1ba9",
"IAX2/[EMAIL PROTECTED]/001516308") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 8ms SCall: 00010 DCall: 0 [213.61.187.157:4569]
VERSION : 2
CALLED NUMBER : 001516308
CALLING NAME: x
LANGUAGE: en
USERNAME: x
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185630951
-- Called [EMAIL PROTECTED]/001516308
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 8ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 1ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
AUTHMETHODS : 3
CHALLENGE : 229696652
USERNAME: x
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00180ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
MD5 RESULT : 8b729ab88c50ba655fef99ef151ad228
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00180ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00171ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
FORMAT : 2
-- Call accepted by 213.61.187.157 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00171ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10009ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10051ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10051ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10009ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10009ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10051ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10934ms SCall: 00306 DCall: 00010 [213.61.187.157:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10934ms SCall: 00010 DCall: 00306 [213.61.187.157:4569]
-- Hungup 'IAX2/voipbuster/10'
== No one is available to answer at this time
-- Executing NoOp("SIP/100-1ba9", "DIALSTATUS=NOANSWER") in new
stack
-- Executing NoOp("SIP/100-1ba9", "HANGUPCAUSE=0") in new stack
-- Executing Congestion("SIP/100-1ba9", "") in new stack
== Spawn extension (internalselections, 9001516308, 6) exited
non-zero on 'SIP/100-1ba9'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Tuesday, August 16, 2005 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
Don Fanning wrote:
> Taking in everyone's suggestions (added a username line also) here is
> what I got.
> Still no joy
> ---
>
> *CLI>
> *CLI>
> *CLI>
> -- Executing SetCallerID("SIP/100-b225", """") in new stack
> -- Executing Dial("SIP/100-b225",
> "IAX2/[EMAIL PROTECTED]/001516308") in new stack
> -- Called [EMAIL PROTECTED]/001516308
> -- Hungup 'IAX2/voipbuster/6'
> == No one is available to answer at this time
> -- Executing Congestion("SIP/100-b225", "") in new stack
> == Spawn extension (internalselections, 9001516308, 3) exited
> non-zero on 'SIP/100-b225'
Put a Noop(HANGUPCAUSE=${HANGUPCAUSE}) and a
Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to
see WHY the call was hungup.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where
[Asterisk-Users] SIP "agent" phone w/ headset
Title: SIP "agent" phone w/ headset I have a call center where we’re looking at converting it from a traditional PBX w/ digital phone “agent” sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we’re currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad, buttons or handset, just a line input and a headset jack. I need the equivalent. I know the first thing you think is why don’t you use the agent’s PC as the VoIP client and do a softphone, however I need to protect the caller from getting cut off should the PC crash/die/etc. While paranoid it’s something where a regular endpoint like an ATA or SIP phone would be the best option. Colin Stefani Tideworks Technology ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and H323 interoperation issue
check the following 1. codecs - is asterisk doing any transcoding ? (i would set both same first) 2. RTP channels. ( the media traffic passes through this channels, make sure your far end and asterisk box has them not obstructed by a NAT or firewall) 3. general phone settings (usually the h323 side) enable debug and see what is the media type. since you are already hearing the ring, it seems to me you are ok with the H323 setup messages and so forth. -apu On 8/16/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hello > > we have one-way audio issue to have Asterisk and H323 work together. We have > SER, Asterisk and H323 module installed. A sip-phone makes the call to a > H323 gateway. The ring and voice can be heard at sip side; there is ring at > far end, but no voice. > > if you have similar issue or clues to address it, please send us a note! > > thank you very much! > steven > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Don Fanning wrote:
Taking in everyone's suggestions (added a username line also) here is
what I got.
Still no joy
---
*CLI>
*CLI>
*CLI>
-- Executing SetCallerID("SIP/100-b225", """") in new stack
-- Executing Dial("SIP/100-b225",
"IAX2/[EMAIL PROTECTED]/001516308") in new stack
-- Called [EMAIL PROTECTED]/001516308
-- Hungup 'IAX2/voipbuster/6'
== No one is available to answer at this time
-- Executing Congestion("SIP/100-b225", "") in new stack
== Spawn extension (internalselections, 9001516308, 3) exited
non-zero on 'SIP/100-b225'
Put a Noop(HANGUPCAUSE=${HANGUPCAUSE}) and a
Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to
see WHY the call was hungup.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.
___
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RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Btw: The number is to my stanaphone DID (so it doesn't bug anyone)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Fanning
Sent: Tuesday, August 16, 2005 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
Taking in everyone's suggestions (added a username line also) here is
what I got.
Still no joy
---
*CLI>
*CLI>
*CLI>
-- Executing SetCallerID("SIP/100-b225", """") in new stack
-- Executing Dial("SIP/100-b225",
"IAX2/[EMAIL PROTECTED]/001516308") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00016ms SCall: 6 DCall: 0 [213.61.187.157:4569]
VERSION : 2
CALLED NUMBER : 001516308
CALLING NAME: x
LANGUAGE: en
USERNAME: x
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185630028
-- Called [EMAIL PROTECTED]/001516308
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00016ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00015ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
AUTHMETHODS : 3
CHALLENGE : 203796716
USERNAME: xx
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00191ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
MD5 RESULT : e682d22660c7a0d278bef6025bcc7dc0
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00191ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00183ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
FORMAT : 2
-- Call accepted by 213.61.187.157 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00183ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
POKE
Timestamp: 00010ms SCall: 7 DCall: 0 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
PONG
Timestamp: 00010ms SCall: 00088 DCall: 7 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00010ms SCall: 7 DCall: 00088 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10017ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
ACK
Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRQ
Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
LAGRP
Timestamp: 10009ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10948ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10948ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
-- Hungup 'IAX2/voipbuster/6'
== No one is available to answer at this time
-- Executing Congestion("SIP/100-b225", "") in new stack
== Spawn extension (internalselections, 9001516308, 3) exited
non-zero on 'SIP/100-b225'
*CLI>
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe
Jensen
Sent: Tuesday, August 16, 2005 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
On 8/16/05, Tony Hoyle <[EMAIL PROTECTED]> wrote:
> Don Fanning wrote:
> >CALLED NUMBER : 1516308
>
> Is that a valid number? AFAIK all voipbuster numbers have to start
> with 0 as there's no local dialing.
Assuming that number is a US number, area code 516, it should be dialed
as 001516308.
Number format at Voipbuster is
00
--
"I am Dyslexic of Borg. Fusistance is retile. Your ass will be
lamitated!"
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://l
RE: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
Taking in everyone's suggestions (added a username line also) here is
what I got.
Still no joy
---
*CLI>
*CLI>
*CLI>
-- Executing SetCallerID("SIP/100-b225", """") in new stack
-- Executing Dial("SIP/100-b225",
"IAX2/[EMAIL PROTECTED]/001516308") in new stack
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00016ms SCall: 6 DCall: 0 [213.61.187.157:4569]
VERSION : 2
CALLED NUMBER : 001516308
CALLING NAME: x
LANGUAGE: en
USERNAME: x
FORMAT : 2
CAPABILITY : 63490
ADSICPE : 2
DATE TIME : 185630028
-- Called [EMAIL PROTECTED]/001516308
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00016ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00015ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
AUTHMETHODS : 3
CHALLENGE : 203796716
USERNAME: xx
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00191ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
MD5 RESULT : e682d22660c7a0d278bef6025bcc7dc0
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00191ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
Timestamp: 00183ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
FORMAT : 2
-- Call accepted by 213.61.187.157 (format gsm)
-- Format for call is gsm
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
Timestamp: 00183ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
POKE
Timestamp: 00010ms SCall: 7 DCall: 0 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
PONG
Timestamp: 00010ms SCall: 00088 DCall: 7 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00010ms SCall: 7 DCall: 00088 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
LAGRQ
Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
LAGRP
Timestamp: 10017ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
ACK
Timestamp: 10017ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass:
LAGRQ
Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass:
LAGRP
Timestamp: 10009ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
ACK
Timestamp: 10009ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
HANGUP
Timestamp: 10948ms SCall: 00325 DCall: 6 [213.61.187.157:4569]
Unknown IE 042 : Present
Ignoring unknown information element 'Unknown IE' (42) of length 1
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass:
ACK
Timestamp: 10948ms SCall: 6 DCall: 00325 [213.61.187.157:4569]
-- Hungup 'IAX2/voipbuster/6'
== No one is available to answer at this time
-- Executing Congestion("SIP/100-b225", "") in new stack
== Spawn extension (internalselections, 9001516308, 3) exited
non-zero on 'SIP/100-b225'
*CLI>
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe
Jensen
Sent: Tuesday, August 16, 2005 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX
connections?
On 8/16/05, Tony Hoyle <[EMAIL PROTECTED]> wrote:
> Don Fanning wrote:
> >CALLED NUMBER : 1516308
>
> Is that a valid number? AFAIK all voipbuster numbers have to start
> with 0 as there's no local dialing.
Assuming that number is a US number, area code 516, it should be dialed
as 001516308.
Number format at Voipbuster is
00
--
"I am Dyslexic of Borg. Fusistance is retile. Your ass will be
lamitated!"
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
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Asterisk-Users@lists.digium.com
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Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
I noticed there isn´t a "username" in your settings, while there is one on Tom´s. Maybe this is the reason for the 'No Authority Found' error ? On 8/16/05, Don Fanning <[EMAIL PROTECTED]> wrote: > Debug below > [voipbuster] > type=peer > host=iax.voipbuster.com > ;host=213.61.187.150 > secret=x > notransfer=yes > context=default > qualify=yes > disallow=all > allow=ulaw > allow=alaw > regards -- Tales Costa [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] realtime caching
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Matthew Boehm > Sent: Tuesday, August 16, 2005 4:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] realtime caching > > > I have reviewed the info below from the sip.sample.conf, but I must be > > dense, still don't get it. > > > > "Do you find the RealTime comments in sip.conf just a little too > confusing? Are you frustrated by the use of double negatives in > configuration options? Do not be afraid. You are not alone. Follow the > path to enlightenment and visit: > http://bugs.digium.com/view.php?id=4075"; > > > It is my understating that removing rtcachefriends will break MWI? Is > > that true? > > Yes. > >What exactly are you trying to accomplish? Are your peers/users not > being updated in your database? Are you sure? Are you watching debug for > SQL log? > > -Matthew > We have a web interface where users can update their dialplan online (not in production yet). The web page modifies the mySQL record. It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
As someone that spent a week or more with anthm refactoring this code I can tell this is how it was when we were done and the code was accepted. So I do know a bit about this area of sip and iax./bOn Aug 16, 2005, at 3:03 PM, Pavel Jezek wrote:I remember many discussions about inteligent codecs negotiation in asterisk, but seems, however, this isn't as simple to implement as it looks... :-( PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quad t1 / 1U rack server combos
I believe that this is a solid state device based trying to follow the principle that the things that would normally go wrong with a server (fans, spinning hard disks, etc) aren't present here and therefore should hang in there longer than a server with a PCI card plugged in. I don't disagree with you though, and would be really happy to see them come out with a dual T1 solution at a lower price point so I could introduce two of these devices across a Quad T1 solution for some kind of redundancy/failover. On 8/16/05, Jonathan k. Creasy <[EMAIL PROTECTED]> wrote: > Wouldn't the point of having a primary and a failover be for redundancy? > Wouldn't purchasing such a device for this configuration void that by > putting you back to a single point of failure? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Cory > Andrews > Sent: Tuesday, August 16, 2005 5:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos > > The foneBRIDGE is not marketed competitively against Digium or Sangoma > 4T1 cards. It is designed specifically for folks who have redundant > Asterisk servers, so that you don't have to purchase a Quad T1 PRI card > for both your primary, and your failovers. > > Cory J Andrews > Partner / Purchasing > +++ > VOIPSupply.com - Everything you need for VOIP > 454 Sonwil Drive > Buffalo, NY 14225 > +++ > tf voice - 800-398-VOIP X22 > l voice - 716.630.1555 X22 > f - 716.630.1548 > e - [EMAIL PROTECTED] > AIM - b2Cory > > > > Damon Estep wrote: > > >I have to agree, 4xT1 density is too low for $2500. If there is some > >magic sauce inside the box then maybe. > > > >What exactly is it? A 4 BRI card in a mini Linux install? Who maintains > >the SIP-ISDN translations? What about docs and support? What are the > >chances the box is really just an mini * server? > > > > > > > > > >>-Original Message- > >>From: [EMAIL PROTECTED] [mailto:asterisk-users- > >>[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy > >>Sent: Tuesday, August 16, 2005 2:51 PM > >>To: Asterisk Users Mailing List - Non-Commercial Discussion > >>Subject: RE: [Asterisk-Users] quad t1 / 1U rack server combos > >> > >>I think the foneBRIDGE is too expensive for what it does. IMHO > >>-jonathan > >> > >>-Original Message- > >>From: [EMAIL PROTECTED] > >>[mailto:[EMAIL PROTECTED] On Behalf Of Cory > >>Andrews > >>Sent: Tuesday, August 16, 2005 4:06 PM > >>To: Asterisk Users Mailing List - Non-Commercial Discussion > >>Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos > >> > >>William - You should take a look at the foneBRIDGE, new product from > >>redFONE. It has (4) PRI interfaces, and you run our to your primary > >> > >> > >and > > > > > >>failover Asterisk servers via Ethernet. It does not do load > >> > >> > >balancing. > > > > > >>but if you have a hardware failure in your primary Asterisk box, you > >> > >> > >can > > > > > >>just fail right over to your secondary box. You don't need any PRI > >>interface cards in your Asterisk host server at all. > >> > >>Cory J Andrews > >>Partner / Purchasing > >>+++ > >>VOIPSupply.com - Everything you need for VOIP > >>454 Sonwil Drive > >>Buffalo, NY 14225 > >>+++ > >>tf voice - 800-398-VOIP X22 > >>l voice - 716.630.1555 X22 > >>f - 716.630.1548 > >>e - [EMAIL PROTECTED] > >>AIM - b2Cory > >> > >> > >> > >>William Boehlke wrote: > >> > >> > >> > >>>In our opinion, BAD idea to put four T1s on a single box, unless you > >>>have another box that also has 4 T1s. > >>> > >>>When, not if, the board fails, you have to take your box down to > >>>replace it. And as with anything having to do with computers you are > >>>guaranteed a failure at a peak time. > >>> > >>>Better to split the load between two boxes. > >>> > >>>William Boehlke > >>>Signate > >>> > >>> > >>> > >>> > >>> > >--- > - > > > > > >>>*From:* [EMAIL PROTECTED] > >>>[mailto:[EMAIL PROTECTED] *On Behalf Of *Chad > >>>Osmond > >>>*Sent:* Tuesday, August 16, 2005 12:15 PM > >>>*To:* Asterisk Users Mailing List - Non-Commercial Discussion > >>>*Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos > >>> > >>>From what I understand (From Sangoma's tech support) and having a > >>> > >>> > >IBM > > > > > >>>x306 SCSI system with an A102u I believe that the system will scale > >>> > >>> > >up > > > > > >>>to 4xT1's easily. > >>>With a full T1 of traffic coming in and playing music on hold, the > >>> > >>> > >CPU > > > > > >>>was at 7% with no transcoding. > >>> > >>>Sangoma cards are supposed to place less draw on the interrupts and > >>>offer some new direct writing to DMA in their A104 cards. You may > >>> > >>> > >want > > > > > >>>to give them a call (Scott or Nenad are the two best people to speak > >>>with). > >>> > >>>From Sangoma README.asterisk: > >>>* Voic
Re: [Asterisk-Users] realtime caching
> I have reviewed the info below from the sip.sample.conf, but I must be > dense, still don’t get it. "Do you find the RealTime comments in sip.conf just a little too confusing? Are you frustrated by the use of double negatives in configuration options? Do not be afraid. You are not alone. Follow the path to enlightenment and visit: http://bugs.digium.com/view.php?id=4075"; > It is my understating that removing rtcachefriends will break MWI? Is > that true? Yes. What exactly are you trying to accomplish? Are your peers/users not being updated in your database? Are you sure? Are you watching debug for SQL log? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 way calling?
Here's the problem - I only have 2 FXO lines. I can call one person, transfer them to the conference room then hang up. Now 1 FXO is used. I can then call another person and transfer them to the conference room and hangup. Now the BOTH lines are used. I can then dial the local extension for the conference room and I have a three way call but that's as far as I get. I've tried establishing a three way call on a single line, but then I cannot seem to transfer the call to the conference room. There's another problem as well. I have three way calling on my Bell lines, so b4 Asterisk, 3 way calls were established by pressing the Flash key and the three way call only used one line. I just discovered that when I do the same thing w/ Asterisk (press the Flash key, dial the other party, press Flash again) BOTH of my POTS lines get used up. Any ideas on what's wrong w/ my configuration? Thanks, Hugh On 8/16/05, Kris Edwards <[EMAIL PROTECTED]> wrote: > As far as quality of meetme, that depends a lot on your setup. I've > had 2 zaps and 3 sips and the quality was perfect. So if you're > talking about 5 people all on zap channels, I think you'll be > satisfied. If you're not, there are a couple of third party apps on > the wiki you could try. > > (for the record, I've never had a meetme conference in which there > were four users on two zap channels 3 wayed, but I suppose * wouldn't > really care about the 3 way part of things... just dump them into the > same room and everyone should be able to talk. (The only thing I can > think of is that the users 3-wayed would not be able to use dtmfs to > control themselves, but rather they would control both users on that > channel.. so, you couldn't have an admin and a user on the same zap > channel, or you couldn't mute just one of the two since * would see > them as the same person)) > > > > On 8/16/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: > > hugolivude wrote: > > > I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). > > > > > > Before I implemented Asterisk, some users were using Bell services to > > > set-up 5 way calling: The user would set up a three way call on one > > > line, switch to the second line, set up another 3 way call and then > > > link the two lines together with the Flash key, thus establishing a 5 > > > way call (the user, 2 others on line 1, another 2 on line 2). How can > > > I accomplish the same thing w/ Asterisk? > > > > You transfer each call into the same MeetMe conference. > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Ita erat quando hic adveni > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hint on parkedcalls
I would like to get hinting to work against extensions used to park calls (ie. call parked on 700, need to be able to exten => 700,hint,/700), and have thus far been unable to find a patch or any information on this functionality. Has anyone done this, or have any input on what would need to be patched to achieve this? -Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New Beta IAX Statistics Program
> Hi, we have put together a small application for Windows to allow you to > check IAX network statistics. The application seems to work fine, but could you point to any information that helps interpret the information that it displays? I've seen the little bit that the new jitterbuffer documentation provides, but I'd like a bit more indepth information on what to do with the raw information. Thanks! Be seeing you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIP signaling vs Media (Voice) Traffic
Greg, Could you elaborate? Thanks, Hugh On 8/11/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > 15 people? Why not use a hosts file? > > Greg > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude > Sent: Thursday, August 11, 2005 3:29 PM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Re: SIP signaling vs Media (Voice) Traffic > > Thanks to JT for the response below: > > The short answer is: no. > > However, there is a solution. Set up a nameserver in your office that > replies with the "internal" address of your Asterisk server to systems > that are on your office LAN, and replies with the "outside" > address of your Asterisk server when asked from hosts outside your LAN. > > This can be done crudely with just launching a local version of BIND > that has a different zone file, or it can be done more elegantly with > DNS "views" on your primary nameserver. I'll let you and Google figure > out how to do it. :-) > > On 8/5/05, hugolivude <[EMAIL PROTECTED]> wrote: > > I have an Asterisk serving 15 people using the X-Lite soft-phone. > > Currently they all register to the internal IP address of Asterisk > > (192.168.1.110). I only use VoIP internally. External calls go PSTN. > > > > I'd like to arrange it so that they register to our external WAN > > address (port forwarded to Asterisk) so that they can go mobile and > > still have Asterisk service. > > > > Is it possible to arrange it so that when in the office, the SIP > > signaling goes through the external WAN, but the Media (Voice) traffic > > > stays local? In other words when a user is on the local LAN, I don't > > want their voice traffic going out on the net and then back in. > > > > Thanks, > > Hugh > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] realtime caching
I could be wrong butsip / iax prune realtime user [user] ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: Tuesday, August 16, 2005 5:10 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] realtime caching Can anyone shed some light on realtime caching? My desired behavior is that MWI works with realtime voicemail/sip/extensions AND updates to the database take place on the next call to the extensions. Right now I have rtcachefriends=yes, and MWI works, but updates to the database for a cached user seem to still require a reload. It is my understating that removing rtcachefriends will break MWI? Is that true? Is there a best of both worlds approach? MWI and realtime updates to extensions? I have reviewed the info below from the sip.sample.conf, but I must be dense, still don’t get it. ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis. ;rtnoupdate=yes ; do not send the update request over realtime. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered when the registration expires ; the friend will vanish from the configuration until requested ; again. If set to an integer, friends expire ; within this number of seconds instead of the ; same as the registration interval ;rtignoreexpire=yes ; when reading a peer from Realtime, if the peer's registration ; has expired based on its registration interval, used the stored ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime caching
Can anyone shed some light on realtime caching? My desired behavior is that MWI works with realtime voicemail/sip/extensions AND updates to the database take place on the next call to the extensions. Right now I have rtcachefriends=yes, and MWI works, but updates to the database for a cached user seem to still require a reload. It is my understating that removing rtcachefriends will break MWI? Is that true? Is there a best of both worlds approach? MWI and realtime updates to extensions? I have reviewed the info below from the sip.sample.conf, but I must be dense, still don’t get it. ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis. ;rtnoupdate=yes ; do not send the update request over realtime. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered when the registration expires ; the friend will vanish from the configuration until requested ; again. If set to an integer, friends expire ; within this number of seconds instead of the ; same as the registration interval ;rtignoreexpire=yes ; when reading a peer from Realtime, if the peer's registration ; has expired based on its registration interval, used the stored ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
Wouldn't the point of having a primary and a failover be for redundancy? Wouldn't purchasing such a device for this configuration void that by putting you back to a single point of failure? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, August 16, 2005 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos The foneBRIDGE is not marketed competitively against Digium or Sangoma 4T1 cards. It is designed specifically for folks who have redundant Asterisk servers, so that you don't have to purchase a Quad T1 PRI card for both your primary, and your failovers. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Damon Estep wrote: >I have to agree, 4xT1 density is too low for $2500. If there is some >magic sauce inside the box then maybe. > >What exactly is it? A 4 BRI card in a mini Linux install? Who maintains >the SIP-ISDN translations? What about docs and support? What are the >chances the box is really just an mini * server? > > > > >>-Original Message- >>From: [EMAIL PROTECTED] [mailto:asterisk-users- >>[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy >>Sent: Tuesday, August 16, 2005 2:51 PM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: RE: [Asterisk-Users] quad t1 / 1U rack server combos >> >>I think the foneBRIDGE is too expensive for what it does. IMHO >>-jonathan >> >>-Original Message- >>From: [EMAIL PROTECTED] >>[mailto:[EMAIL PROTECTED] On Behalf Of Cory >>Andrews >>Sent: Tuesday, August 16, 2005 4:06 PM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos >> >>William - You should take a look at the foneBRIDGE, new product from >>redFONE. It has (4) PRI interfaces, and you run our to your primary >> >> >and > > >>failover Asterisk servers via Ethernet. It does not do load >> >> >balancing. > > >>but if you have a hardware failure in your primary Asterisk box, you >> >> >can > > >>just fail right over to your secondary box. You don't need any PRI >>interface cards in your Asterisk host server at all. >> >>Cory J Andrews >>Partner / Purchasing >>+++ >>VOIPSupply.com - Everything you need for VOIP >>454 Sonwil Drive >>Buffalo, NY 14225 >>+++ >>tf voice - 800-398-VOIP X22 >>l voice - 716.630.1555 X22 >>f - 716.630.1548 >>e - [EMAIL PROTECTED] >>AIM - b2Cory >> >> >> >>William Boehlke wrote: >> >> >> >>>In our opinion, BAD idea to put four T1s on a single box, unless you >>>have another box that also has 4 T1s. >>> >>>When, not if, the board fails, you have to take your box down to >>>replace it. And as with anything having to do with computers you are >>>guaranteed a failure at a peak time. >>> >>>Better to split the load between two boxes. >>> >>>William Boehlke >>>Signate >>> >>> >>> >>> >>> >--- - > > >>>*From:* [EMAIL PROTECTED] >>>[mailto:[EMAIL PROTECTED] *On Behalf Of *Chad >>>Osmond >>>*Sent:* Tuesday, August 16, 2005 12:15 PM >>>*To:* Asterisk Users Mailing List - Non-Commercial Discussion >>>*Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos >>> >>>From what I understand (From Sangoma's tech support) and having a >>> >>> >IBM > > >>>x306 SCSI system with an A102u I believe that the system will scale >>> >>> >up > > >>>to 4xT1's easily. >>>With a full T1 of traffic coming in and playing music on hold, the >>> >>> >CPU > > >>>was at 7% with no transcoding. >>> >>>Sangoma cards are supposed to place less draw on the interrupts and >>>offer some new direct writing to DMA in their A104 cards. You may >>> >>> >want > > >>>to give them a call (Scott or Nenad are the two best people to speak >>>with). >>> >>>From Sangoma README.asterisk: >>>* Voice data is channelized and grouped into 8 byte chunks in >>>HARDWARE. Each voice channel is then DMAed directly into the >>>ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, >>>resulting in better performance and scalability.* >>> >>> >>>It sounds to me like that would be once advantage over Digiums >>> >>> >cards. > > >>>They also have Hardware PRI functions that are passed directly to >>> >>> >>libpri. >> >> >>>http://sangoma.com/linux/README.asterisk >>> >>>Hope that helps. >>> >>>Chad >>> >>> >>> >>> >--- - > > >>>*From:* [EMAIL PROTECTED] >>>[mailto:[EMAIL PROTECTED] *On Behalf Of >>> >>> >*Damon > > >>>Estep >>>*Sent:* August 16, 2005 12:33 PM >>>*To:* asterisk-users@lists.digium.com >>>*Sub
Re: [Asterisk-Users] quad t1 / 1U rack server combos
The foneBRIDGE is not marketed competitively against Digium or Sangoma 4T1 cards. It is designed specifically for folks who have redundant Asterisk servers, so that you don't have to purchase a Quad T1 PRI card for both your primary, and your failovers. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Damon Estep wrote: I have to agree, 4xT1 density is too low for $2500. If there is some magic sauce inside the box then maybe. What exactly is it? A 4 BRI card in a mini Linux install? Who maintains the SIP-ISDN translations? What about docs and support? What are the chances the box is really just an mini * server? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Tuesday, August 16, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] quad t1 / 1U rack server combos I think the foneBRIDGE is too expensive for what it does. IMHO -jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, August 16, 2005 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos William - You should take a look at the foneBRIDGE, new product from redFONE. It has (4) PRI interfaces, and you run our to your primary and failover Asterisk servers via Ethernet. It does not do load balancing. but if you have a hardware failure in your primary Asterisk box, you can just fail right over to your secondary box. You don't need any PRI interface cards in your Asterisk host server at all. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory William Boehlke wrote: In our opinion, BAD idea to put four T1s on a single box, unless you have another box that also has 4 T1s. When, not if, the board fails, you have to take your box down to replace it. And as with anything having to do with computers you are guaranteed a failure at a peak time. Better to split the load between two boxes. William Boehlke Signate *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad Osmond *Sent:* Tuesday, August 16, 2005 12:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, the CPU was at 7% with no transcoding. Sangoma cards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: * Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability.* It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon Estep *Sent:* August 16, 2005 12:33 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the
RE: [Asterisk-Users] quad t1 / 1U rack server combos
I have to agree, 4xT1 density is too low for $2500. If there is some magic sauce inside the box then maybe. What exactly is it? A 4 BRI card in a mini Linux install? Who maintains the SIP-ISDN translations? What about docs and support? What are the chances the box is really just an mini * server? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy > Sent: Tuesday, August 16, 2005 2:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] quad t1 / 1U rack server combos > > I think the foneBRIDGE is too expensive for what it does. IMHO > -jonathan > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Cory > Andrews > Sent: Tuesday, August 16, 2005 4:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos > > William - You should take a look at the foneBRIDGE, new product from > redFONE. It has (4) PRI interfaces, and you run our to your primary and > > failover Asterisk servers via Ethernet. It does not do load balancing. > but if you have a hardware failure in your primary Asterisk box, you can > > just fail right over to your secondary box. You don't need any PRI > interface cards in your Asterisk host server at all. > > Cory J Andrews > Partner / Purchasing > +++ > VOIPSupply.com - Everything you need for VOIP > 454 Sonwil Drive > Buffalo, NY 14225 > +++ > tf voice - 800-398-VOIP X22 > l voice - 716.630.1555 X22 > f - 716.630.1548 > e - [EMAIL PROTECTED] > AIM - b2Cory > > > > William Boehlke wrote: > > > In our opinion, BAD idea to put four T1s on a single box, unless you > > have another box that also has 4 T1s. > > > > When, not if, the board fails, you have to take your box down to > > replace it. And as with anything having to do with computers you are > > guaranteed a failure at a peak time. > > > > Better to split the load between two boxes. > > > > William Boehlke > > Signate > > > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad > > Osmond > > *Sent:* Tuesday, August 16, 2005 12:15 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos > > > > From what I understand (From Sangoma's tech support) and having a IBM > > x306 SCSI system with an A102u I believe that the system will scale up > > > to 4xT1's easily. > > With a full T1 of traffic coming in and playing music on hold, the CPU > > > was at 7% with no transcoding. > > > > Sangoma cards are supposed to place less draw on the interrupts and > > offer some new direct writing to DMA in their A104 cards. You may want > > > to give them a call (Scott or Nenad are the two best people to speak > > with). > > > > From Sangoma README.asterisk: > > * Voice data is channelized and grouped into 8 byte chunks in > > HARDWARE. Each voice channel is then DMAed directly into the > > ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, > > resulting in better performance and scalability.* > > > > > > It sounds to me like that would be once advantage over Digiums cards. > > They also have Hardware PRI functions that are passed directly to > libpri. > > http://sangoma.com/linux/README.asterisk > > > > Hope that helps. > > > > Chad > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon > > Estep > > *Sent:* August 16, 2005 12:33 PM > > *To:* asterisk-users@lists.digium.com > > *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos > > > > It is amazing to me at this point that there is not an official Digium > > > list of supported servers (including 1u models!). Clearly the number 1 > > > issue with the Digium PRI cards is the server that they are used in. > > > > > > > > The new cards even go as far as listing server that DO NOT work on the > > > Digium site! > > > > > > > > The wiki references are old and do not have any testing parameters. > > > > > > > > C'mon guys! Certify a few current model servers and be done with it. > > > > > > > > Without that information I must again ask the question; > > > > > > > > What 1u server combos work with the new quad pri cards UNDER LOAD > > (more than 75% channel use). Every user that buys a Digium PRI card > > should not have to play hit or miss with 2 or 3 servers that cost more > > > than the card to get it to work. > > > > > > > > Please Please Please publish something useful to support the sale of > > PRI cards. > > > > > > > > Damon > > > > > > -- > > No virus found in this incoming message. > > Checked by AVG Anti-Virus. > > Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: > 8/15/2005 > > > >
RE: [Asterisk-Users] Called Party Identification on Polycom IP501
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Anthony Rodgers > Sent: Tuesday, August 16, 2005 1:21 PM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Called Party Identification on Polycom IP501 > > Greetings, > > The Polycom SIP 1.5 Admin Guide says this: > > "3.1.8 Connected Party Identification > > Where possible, the identity of the remote party to which the user has > connected is displayed and logged. The connected party identity is > derived from the network signaling. In some cases the remote party > will be different from the called party identity due to network call > diversion." > > Does anyone know if * can provide the "network signaling" required? If > so, how? > > Regards, > -- > Anthony Rodgers > Business Systems Analyst > District of North Vancouver > Web: http://www.dnv.org > RSS Feed: http://www.dnv.org/rss.asp > That is very dependent on how the call egresses from *, ISDN, POTS, SIP, ??? Who are you calling? If I recall correctly it will work when you call another extension on the * box, but the signaling for that info does not exists in PRI/T1/POTS, so it is not an * issue if you area calling out, * cant get the info from the telco, so * cant send it to the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
I think the foneBRIDGE is too expensive for what it does. IMHO -jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, August 16, 2005 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos William - You should take a look at the foneBRIDGE, new product from redFONE. It has (4) PRI interfaces, and you run our to your primary and failover Asterisk servers via Ethernet. It does not do load balancing. but if you have a hardware failure in your primary Asterisk box, you can just fail right over to your secondary box. You don't need any PRI interface cards in your Asterisk host server at all. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory William Boehlke wrote: > In our opinion, BAD idea to put four T1s on a single box, unless you > have another box that also has 4 T1s. > > When, not if, the board fails, you have to take your box down to > replace it. And as with anything having to do with computers you are > guaranteed a failure at a peak time. > > Better to split the load between two boxes. > > William Boehlke > Signate > > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad > Osmond > *Sent:* Tuesday, August 16, 2005 12:15 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos > > From what I understand (From Sangoma's tech support) and having a IBM > x306 SCSI system with an A102u I believe that the system will scale up > to 4xT1's easily. > With a full T1 of traffic coming in and playing music on hold, the CPU > was at 7% with no transcoding. > > Sangoma cards are supposed to place less draw on the interrupts and > offer some new direct writing to DMA in their A104 cards. You may want > to give them a call (Scott or Nenad are the two best people to speak > with). > > From Sangoma README.asterisk: > * Voice data is channelized and grouped into 8 byte chunks in > HARDWARE. Each voice channel is then DMAed directly into the > ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, > resulting in better performance and scalability.* > > > It sounds to me like that would be once advantage over Digiums cards. > They also have Hardware PRI functions that are passed directly to libpri. > http://sangoma.com/linux/README.asterisk > > Hope that helps. > > Chad > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon > Estep > *Sent:* August 16, 2005 12:33 PM > *To:* asterisk-users@lists.digium.com > *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos > > It is amazing to me at this point that there is not an official Digium > list of supported servers (including 1u models!). Clearly the number 1 > issue with the Digium PRI cards is the server that they are used in. > > > > The new cards even go as far as listing server that DO NOT work on the > Digium site! > > > > The wiki references are old and do not have any testing parameters. > > > > C'mon guys! Certify a few current model servers and be done with it. > > > > Without that information I must again ask the question; > > > > What 1u server combos work with the new quad pri cards UNDER LOAD > (more than 75% channel use). Every user that buys a Digium PRI card > should not have to play hit or miss with 2 or 3 servers that cost more > than the card to get it to work. > > > > Please Please Please publish something useful to support the sale of > PRI cards. > > > > Damon > > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 > > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 > >--- - > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
[Asterisk-Users] RE:Asterisk-Users] PhoneCALL v2.6.1 - Released
Really great job, it looks like exactly what we were searching for, since get started with asterisk. Keep on going with this excellent work. Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
What does the foneBRIDGE do that a Lucent TNT won't? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Cory Andrews > Sent: Tuesday, August 16, 2005 2:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos > > William - You should take a look at the foneBRIDGE, new product from > redFONE. It has (4) PRI interfaces, and you run our to your primary and > failover Asterisk servers via Ethernet. It does not do load balancing. > but if you have a hardware failure in your primary Asterisk box, you can > just fail right over to your secondary box. You don't need any PRI > interface cards in your Asterisk host server at all. > > Cory J Andrews > Partner / Purchasing > +++ > VOIPSupply.com - Everything you need for VOIP > 454 Sonwil Drive > Buffalo, NY 14225 > +++ > tf voice - 800-398-VOIP X22 > l voice - 716.630.1555 X22 > f - 716.630.1548 > e - [EMAIL PROTECTED] > AIM - b2Cory > > > > William Boehlke wrote: > > > In our opinion, BAD idea to put four T1s on a single box, unless you > > have another box that also has 4 T1s. > > > > When, not if, the board fails, you have to take your box down to > > replace it. And as with anything having to do with computers you are > > guaranteed a failure at a peak time. > > > > Better to split the load between two boxes. > > > > William Boehlke > > Signate > > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad > > Osmond > > *Sent:* Tuesday, August 16, 2005 12:15 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos > > > > From what I understand (From Sangoma's tech support) and having a IBM > > x306 SCSI system with an A102u I believe that the system will scale up > > to 4xT1's easily. > > With a full T1 of traffic coming in and playing music on hold, the CPU > > was at 7% with no transcoding. > > > > Sangoma cards are supposed to place less draw on the interrupts and > > offer some new direct writing to DMA in their A104 cards. You may want > > to give them a call (Scott or Nenad are the two best people to speak > > with). > > > > From Sangoma README.asterisk: > > * Voice data is channelized and grouped into 8 byte chunks in > > HARDWARE. Each voice channel is then DMAed directly into the > > ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, > > resulting in better performance and scalability.* > > > > > > It sounds to me like that would be once advantage over Digiums cards. > > They also have Hardware PRI functions that are passed directly to > libpri. > > http://sangoma.com/linux/README.asterisk > > > > Hope that helps. > > > > Chad > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon > > Estep > > *Sent:* August 16, 2005 12:33 PM > > *To:* asterisk-users@lists.digium.com > > *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos > > > > It is amazing to me at this point that there is not an official Digium > > list of supported servers (including 1u models!). Clearly the number 1 > > issue with the Digium PRI cards is the server that they are used in. > > > > > > > > The new cards even go as far as listing server that DO NOT work on the > > Digium site! > > > > > > > > The wiki references are old and do not have any testing parameters. > > > > > > > > C'mon guys! Certify a few current model servers and be done with it. > > > > > > > > Without that information I must again ask the question; > > > > > > > > What 1u server combos work with the new quad pri cards UNDER LOAD > > (more than 75% channel use). Every user that buys a Digium PRI card > > should not have to play hit or miss with 2 or 3 servers that cost more > > than the card to get it to work. > > > > > > > > Please Please Please publish something useful to support the sale of > > PRI cards. > > > > > > > > Damon > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
> > > Are you saying realtime mysql is not clever? That is exactly what it is > > supposed to do. > > > > > BTW, how do you integrate mysql with asterisk? > any link, documention, tutorials would be greatly helpful. > Search www.voip-info.org for asterisk realtime ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What should my next steps in troubleshooting this TDM04B error be?
Hello I have installed a TDM04B and disabled any devices not required in my PC. (TDM04B is analog card with 4 ports to plug into telephone co lines). I am running this version of * Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on a i686 running Linux As you see below the wctdm module is loaded: pbx root # lsmod Module Size Used by binfmt_misc 12296 1 - Live 0xde839000 wctdm 129216 0 - Live 0xde855000 zaptel 235844 1 wctdm, Live 0xde877000 hdlc 24576 1 zaptel, Live 0xde84e000 syncppp 17116 1 hdlc, Live 0xde848000 ppp_generic 30612 1 zaptel, Live 0xde83f000 slhc 7808 1 ppp_generic, Live 0xde829000 crc_ccitt 2432 1 zaptel, Live 0xde806000 via_rhine 21252 0 - Live 0xde82d000 mii 5120 1 via_rhine, Live 0xde81d000 crc32 4608 1 via_rhine, Live 0xde81a000 rtc 12748 0 - Live 0xde82 But running ztfcg gives me this error: pbx root # ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) What does this mean exactly? Is it saying it can't find the hardware? Then get these errors loading Asterisk: == Parsing '/etc/asterisk/zapata.conf': Found Aug 16 19:56:12 WARNING[363]: chan_zap.c:792 zt_open: Unable to specify channel 1: No such device or address Aug 16 19:56:12 ERROR[363]: chan_zap.c:6327 mkintf: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 Aug 16 19:56:12 ERROR[363]: chan_zap.c:9337 setup_zap: Unable to register channel '1' Aug 16 19:56:12 WARNING[363]: loader.c:396 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Zap' Aug 16 19:56:12 WARNING[363]: loader.c:501 load_modules: Loading module chan_zap.so failed! My zaptel.conf: fxsks=1-4 loadzone = uk defaultzone = uk My zapata.conf (abbreviated): [channels] context=default group=1 signalling=fxs_ks channel => 1-4 What do I need to look at next? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
> Are you saying realtime mysql is not clever? That is exactly what it is > supposed to do. > BTW, how do you integrate mysql with asterisk? any link, documention, tutorials would be greatly helpful. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quad t1 / 1U rack server combos
William - You should take a look at the foneBRIDGE, new product from redFONE. It has (4) PRI interfaces, and you run our to your primary and failover Asterisk servers via Ethernet. It does not do load balancing. but if you have a hardware failure in your primary Asterisk box, you can just fail right over to your secondary box. You don't need any PRI interface cards in your Asterisk host server at all. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory William Boehlke wrote: In our opinion, BAD idea to put four T1s on a single box, unless you have another box that also has 4 T1s. When, not if, the board fails, you have to take your box down to replace it. And as with anything having to do with computers you are guaranteed a failure at a peak time. Better to split the load between two boxes. William Boehlke Signate *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Chad Osmond *Sent:* Tuesday, August 16, 2005 12:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* RE: [Asterisk-Users] quad t1 / 1U rack server combos From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, the CPU was at 7% with no transcoding. Sangoma cards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: * Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability.* It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon Estep *Sent:* August 16, 2005 12:33 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C’mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
I remember many discussions about inteligent codecs negotiation in asterisk, but seems, however, this isn't as simple to implement as it looks... :-( PJ Erik Versaevel wrote: That should be controllable by a weight, for example 2 peers: A --> G729, G711 B --> G711, G729 What's currently happening is that * starts transcoding between the two (g729 for A and G711 for B), what i would like is to apply a weight to peer A so that the codec of choise at both sides becomes the preffered choise of A (G729) on both sides so there won't be any transcoding. This would allow for some nice things as fax passtrough (A and B has to use G711 then, but if the weigted A says G711, B would use G711 to). Kind regards, Erik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
> Damon Estep wrote: > > > What 1u server combos work with the new quad pri cards UNDER LOAD (more > > than 75% channel use). Every user that buys a Digium PRI card should not > > have to play hit or miss with 2 or 3 servers that cost more than the > > card to get it to work. > > We use a Sangoma 4 port T1 card in our Dell Poweredge 1850 (1U) and it > works like a champ. > > -Matthew > One of the obvious disadvantages in using Sangoma cards would be Marksters interest is supporting them, using a TNT right now, and there are minor caller ID issues. The whole idea is to use a card offered by the company managing the project so interoperability is almost guaranteed. With that aside, what are the other pros/cons of the sagnoma cards? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
> How about this: > > 1. Put all the routes of all the providers in a MySQL table > 2. Write a script with a 'clever' algorithm to find out cheapest route of > each prefix. Are you saying realtime mysql is not clever? That is exactly what it is supposed to do. > 3. Based on #2.. make a lcr_cheapest_route.conf > 4. include lcr_cheapest_route.conf in extension.conf That is what we are doing, 3 minute reloads! > > But I don't know, how much resource asterisk will take after loading > lcr_cheapest_route.conf > Also, I don't have any idea about the performance would be. > > What do you think? > > Thanks > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > Sent: Tue, 16 Aug 2005 12:57:14 -0400 > > To: asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] Asterisk and LCR > > > > On Tue, Aug 16, 2005 at 10:22:01AM -0600, Damon Estep wrote: > > > > > > Any input from others that have already done what I am doing would be > > > helpful, what works best? > > > > For 100k routes+, you will have trouble holding them in a SQL database, > > particularly if your route selection query is complex. With a modern PC > > running PostgreSQL, you'll run into trouble at around 250k BHCA even > with > > a much smaller number of routes. (This is quite apart from Asterisk > > itself, > > try writing a simple program that runs sample queries in a loop, perhaps > > with several threads. To a certain extent it depends on how you write > the > > query and how judiciously you place indexes on the tables) When you want > > NPANXX granularity from several carriers (commonly 75-100k routes each) > > you'll get hit even worse. > > > > In my experience the safe limits of this approach are about a 2x DS3 > > worth of traffic with 10,000 routes in the table... After that you've > got > > to pull everything into RAM and write a clever route selection > > algorithm... > > > > -w > > -- > > William Waites > > ww [EMAIL PROTECTED] magicphone.ca > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk- > users___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
On 8/16/05, Tony Hoyle <[EMAIL PROTECTED]> wrote: > Don Fanning wrote: > >CALLED NUMBER : 1516308 > > Is that a valid number? AFAIK all voipbuster numbers have to start with > 0 as there's no local dialing. Assuming that number is a US number, area code 516, it should be dialed as 001516308. Number format at Voipbuster is 00 -- "I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated!" ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TxFax -> RxFax on same machine hangs
Hi, I noticed a strange behaviour: Faxing using spandsp (TxFax) from my asterisk box to my old, common fax machine at home works fine. Faxing from the same box to my office pc-fax (Hylafax) also worke fine. Receveiving faxes on my asterisk box using spandsp (RxFax) also works fine. It is a PSTN number connected to the digium card of that asterisk box. Then I faxed from my asterisk box (TxFax) to that PSTN number, my asterisk box answers the call, TxFax and RxFax, both start, but no tif file is created in the directory, where normally the fax files are created. Using show channel I can still see both apps, TxFax and RxFax, even after half an hour. Then I stopped using soft hangup, and tried again several times: Same result. So, do I have to avoid to fax to myself in any case, because I risk to produce those never terminating jobs, which probably consume some resources? How can I enable asterisk to fax to itsself? Well, it won't be the normal operation, but when allowing clients to fax, it can happen by chance, that someone faxes to another user on the same machine without knowing it. Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
In our opinion, BAD idea to put four T1s on a single box, unless you have another box that also has 4 T1s. When, not if, the board fails, you have to take your box down to replace it. And as with anything having to do with computers you are guaranteed a failure at a peak time. Better to split the load between two boxes. William Boehlke Signate From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad OsmondSent: Tuesday, August 16, 2005 12:15 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] quad t1 / 1U rack server combos From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, the CPU was at 7% with no transcoding. Sangoma cards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability. It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: August 16, 2005 12:33 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C’mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Called Party Identification on Polycom IP501
Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion." Does anyone know if * can provide the "network signaling" required? If so, how? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 way calling?
As far as quality of meetme, that depends a lot on your setup. I've had 2 zaps and 3 sips and the quality was perfect. So if you're talking about 5 people all on zap channels, I think you'll be satisfied. If you're not, there are a couple of third party apps on the wiki you could try. (for the record, I've never had a meetme conference in which there were four users on two zap channels 3 wayed, but I suppose * wouldn't really care about the 3 way part of things... just dump them into the same room and everyone should be able to talk. (The only thing I can think of is that the users 3-wayed would not be able to use dtmfs to control themselves, but rather they would control both users on that channel.. so, you couldn't have an admin and a user on the same zap channel, or you couldn't mute just one of the two since * would see them as the same person)) On 8/16/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: > hugolivude wrote: > > I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). > > > > Before I implemented Asterisk, some users were using Bell services to > > set-up 5 way calling: The user would set up a three way call on one > > line, switch to the second line, set up another 3 way call and then > > link the two lines together with the Flash key, thus establishing a 5 > > way call (the user, 2 others on line 1, another 2 on line 2). How can > > I accomplish the same thing w/ Asterisk? > > You transfer each call into the same MeetMe conference. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ita erat quando hic adveni ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quad t1 / 1U rack server combos
From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, the CPU was at 7% with no transcoding. Sangoma cards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability. It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: August 16, 2005 12:33 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C’mon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5 way calling?
Hugh, Meetme works great, yes you should definitely do this. One of the main reasons I use asterisk at home is for running many conference calls each day. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of hugolivude > Sent: Tuesday, 16 August 2005 2:41 PM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] 5 way calling? > > I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). > > Before I implemented Asterisk, some users were using Bell services to > set-up 5 way calling: The user would set up a three way call on one > line, switch to the second line, set up another 3 way call and then > link the two lines together with the Flash key, thus establishing a 5 > way call (the user, 2 others on line 1, another 2 on line 2). How can > I accomplish the same thing w/ Asterisk? > > I had thought that perhaps the user could set up a three way call, > transfer that to a Meet-Me conference room, then set up another three > way call on the other line and transfer that to the same Meet-Me > conference room. Would that work? > > It's a little difficult for me to "try" this, so if you have insight > I'd be grateful. BTW what's the quality like in the Meet-Me > conference room? Is it comparable to what Bell would provide? > > Thanks, > Hugh > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and H323 interoperation issue
Hello we have one-way audio issue to have Asterisk and H323 work together. We have SER, Asterisk and H323 module installed. A sip-phone makes the call to a H323 gateway. The ring and voice can be heard at sip side; there is ring at far end, but no voice. if you have similar issue or clues to address it, please send us a note! thank you very much! steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 109
These hardware issues are a joke. Go with Sangoma- they immplement PCI properly and don't have any of these system board issues. One of their sales guys told me that they would take their prodcut back and pay me $500 if I found a PCI system newer than 2 years that they did not work with. Pretty compelling no?Pat Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax
Brian West wrote: No it is really about faxing. As someone that has first hand knowledge of the case outlined on Groklaw, it is in fact about faxing. Go read the two patents very carefully! If you email it you break 638, if you store it you break 021. How, then, do these patents themselves not violate Brooktrout's own portfolio of earlier fax-specific patents covering virtually the same things? Lee. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 501 dialing problem
When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 way calling?
hugolivude wrote: I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How can I accomplish the same thing w/ Asterisk? You transfer each call into the same MeetMe conference. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problems with eyebeam - video phone
Hi, I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I only use H.263 and SIP. (G.729) Now, the more important question is if you register on the domain on the Eyebeam software. I found that this was the full secret about this. Let me know your configuration on the Eyebeam side. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 11:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problems with eyebeam - video phone I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ; allow=h263p ; H.263p is the enhanced video codec context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf And I left only H.263 basic in codec's configuration in Video Phone. No chance to get the communication in H.263 protocol. I saw that to use H.263+ protocol I need Asterisk CVS. I am not using asterisk CVS I am using asterisk 1.0.9 (last stable version a couple of week ago..) Is there any chance to make asterisk 1.0.9 to support SIP video calls in eyeBeam ? Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax
Although Groklaw seems to think that these suits are about faxing, I don't think that they really are. See: http://www.hylafax.org/archive/2005-08/msg00107.htmlLee.No it is really about faxing. As someone that has first hand knowledge of the case outlined on Groklaw, it is in fact about faxing.Go read the two patents very carefully! If you email it you break 638, if you store it you break 021./b___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] quad t1 / 1U rack server combos
Damon, Here is our offering which we have extensively tested. It is VERY capable handling all 96/120 channels at once, and there are no issues between the quad card and the motherboard. We have a great number of customers running this configuration. http://www.govarion.com/product_info.php?products_id=11 Best Regards, Ben Bawkon - Original Message - From: Andrew Latham To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [Asterisk-Users] quad t1 / 1U rack server combos Sent: 8/16/2005 1:36:45 PM My personal fav. http://www.provantage.com/buy-7ASUN00P-asus-networking-1u-p4-bbns-e7210-52x-cd-2x64-2sata-h-2geth-300w-ap140r-e1-aa2-shopping.htm On 8/16/05, William Lloyd <[EMAIL PROTECTED]> wrote: > I use this box with no problems at all. > > http://www.tyan.com/products/html/gx28b2881.html > > -bill > > On 16-Aug-05, at 12:32 PM, Damon Estep wrote: > > > It is amazing to me at this point that there is not an official > > Digium list of supported servers (including 1u models!). Clearly > > the number 1 issue with the Digium PRI cards is the server that > > they are used in. > > > > > > The new cards even go as far as listing server that DO NOT work on > > the Digium site! > > > > > > The wiki references are old and do not have any testing parameters. > > > > > > C'mon guys! Certify a few current model servers and be done with it. > > > > > > Without that information I must again ask the question; > > > > > > What 1u server combos work with the new quad pri cards UNDER LOAD > > (more than 75% channel use). Every user that buys a Digium PRI card > > should not have to play hit or miss with 2 or 3 servers that cost > > more than the card to get it to work. > > > > > > Please Please Please publish something useful to support the sale > > of PRI cards. > > > > > > Damon > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.govarion.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 5 way calling?
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How can I accomplish the same thing w/ Asterisk? I had thought that perhaps the user could set up a three way call, transfer that to a Meet-Me conference room, then set up another three way call on the other line and transfer that to the same Meet-Me conference room. Would that work? It's a little difficult for me to "try" this, so if you have insight I'd be grateful. BTW what's the quality like in the Meet-Me conference room? Is it comparable to what Bell would provide? Thanks, Hugh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail file permissions
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I'd like to give my Asterisk users the option of cleaning up their voicemail mailbox from their Windows PCs. I set up Samba and added all the users with restricted access to their mailbox only, but here's the problem: The voicemail .wav files that Asterisk creates have root as both owner and group. Since the users do not have root privileges, they can't do much with the files. BTW I'm not sure why the voicemail .wav files have root as both owner and group because I followed the instructions for running Asterisk other than root (see http://www.voip-info.org/wiki-Asterisk+non-root). Is there a way around this w/o giving everyone root privileges! Thanks, Hugh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MFC/R2 DTMF and digits "*" and "#"
Hi all ,i can configure MFC/R2 but i can´t send "*" and "#" digits using DTMF . but others digits works well i receive and make calls. thks for you attentions in annex this my conf files zaptel.conf Description: Binary data unicall.conf Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP channels not cleared
Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.1730022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.1730022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.1730016513973 234f7bba140 00102/00103 unknow(d) Rx: BYE 10.223.51.1730027226765 487b770b231 00102/00103 unknow(d) Rx: BYE 10.223.51.1730016513973 69b59aa2084 00102/00103 unknow(d) Rx: BYE 10.223.51.1730199820127 60ef984904a 00102/00103 unknow(d) Rx: BYE 10.223.51.1730081805135 45bf3e8c287 00102/00103 unknow(d) Rx: BYE I have thousands of them in 'sip show channels' and is increasing but it only shows 50 calls in 'show channels'. I believe this eats up memory. Sooner or later my system will run out of memory or get the 'Too many file opened' error. I have made a sip trace on asterisk and seems like they all share a same SIP message flow. When asterisk send an INVITE to other sip server say B. B will reply with Trying. When B found out that the actual destination can not be reached, it sends a BYE to asterisk. Asterisk then reply with a 200 OK. Call is hangup succesfully but 'sip show channels' still list the call record and never go away untill asterisk is restart. See below: Aug 15 18:35:32 VERBOSE[12402] logger.c: Reliably Transmitting (no NAT) to 10.223.51.173:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0^M Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4^M From: "DADAS" ;tag=as64c4813c^M To: ^M Contact: ^M Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Date: Mon, 15 Aug 2005 10:35:32 GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY^M Content-Type: application/sdp^M Content-Length: 160^M ^M v=0^M o=root 12402 12402 IN IP4 10.21.99.221^M s=session^M c=IN IP4 10.21.99.221^M t=0 0^M m=audio 10986 RTP/AVP 8^M a=rtpmap:8 PCMA/8000^M a=silenceSupp:off - - - -^M Aug 15 18:35:32 VERBOSE[15229] logger.c: <-- SIP read from 10.223.51.173:5060: SIP/2.0 100 Trying Call-Id: [EMAIL PROTECTED] CSeq: 102 INVITE From: "DADAS" ;tag=as64c4813c To: Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4 Aug 15 18:35:39 VERBOSE[15229] logger.c: <-- SIP read from 10.223.51.173:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Call-Id: [EMAIL PROTECTED] Content-Length: 0 CSeq: 103 BYE From: ;tag=a10111834662596 To: "DADAS" ;tag=as64c4813c Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33 Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4 Aug 15 18:35:39 VERBOSE[15229] logger.c: Transmitting (no NAT) to 10.223.51.173:5060: SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33^M Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4^M From: ;tag=a10111834662596^M To: "DADAS" ;tag=as64c4813c^M Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY^M Contact: ^M Content-Length: 0^M The SIP message exchange seems to be comply to the standard. Is this a bug in asterisk? I have a system where there is always call going on and I cant schedule asterisk to be restarted at any time to clear the channels. Any idea? I have CVS HEAD runnung on fedora 3. Thanks CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Maximum remote directory size in Polycom IP501
I managed to trace this to a bad record in the data - as pointed out on the list, the phone can easily handle a 60K file, provided it contains valid XML! Another win for Polycom. A. On Aug 15, 2005, at 8:26 PM, Anthony Rodgers wrote: Greetings, We are trying to make our corporate directory (around 400 entries) available via TFTP to some Polycom IP501 phones. A small (~40 entries or so) file works, but the full file fails to load. Does anyone know what the upper limit on directory entries is? The size of the XML file itself is only 60K - you'd think that would all fit into the phone with no problems. I would appreciate a cc: on any replies to the list, as I don't always get to read it in a timely fashion. thanks! Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax
Brian West wrote: Just an FYI http://www.groklaw.net/article.php?story=2005080914234645 Although Groklaw seems to think that these suits are about faxing, I don't think that they really are. See: http://www.hylafax.org/archive/2005-08/msg00107.html Lee. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] adding another fxo card
Ric Moseley wrote: Also, what does the RED mean in this? [EMAIL PROTECTED]:~]#more /proc/zaptel/* :: /proc/zaptel/1 :: Span 1: WCFXO/0 "Generic Clone Board 1" 1 WCFXO/0/0 FXSKS (In use) :: /proc/zaptel/2 :: Span 2: WCFXO/1 "Generic Clone Board 2" RED 2 WCFXO/1/0 FXSKS (In use) It means you don't have a phone line plugged into the card. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incompatible destination (88) Error Message
George K. Konstantoulakis wrote: Geia sou Irakli, I would have to agree with Bryce that from the debug output the problem seems to be with the dialed number. "Unkown Number Type" & "Unkown Number plan" point to that. You should probably check out if you can start extensions with 3 ... Maybe he needs pridialplan=unknown ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and LCR
On Tue, Aug 16, 2005 at 09:29:24AM -0800, Innocent Evil wrote: > How about this: > > 1. Put all the routes of all the providers in a MySQL table > 2. Write a script with a 'clever' algorithm to find out cheapest route of > each prefix. > 3. Based on #2.. make a lcr_cheapest_route.conf > 4. include lcr_cheapest_route.conf in extension.conf > > But I don't know, how much resource asterisk will take after loading > lcr_cheapest_route.conf > Also, I don't have any idea about the performance would be. That would probably work, but there are limitations with this approach: - changing the routes (which might happen more than you might expect, especially if you keep real-time metrics) will mean doing an "extensions reload" which is, apparently, disruptive if it takes any appreciable amount of time. - it means that (short of doubling or tripling the size of your tables) you are restricted to one and only one route per destination, which may or may not be what you want. - having multiple contexts (aka policies) also means doubling/tripling/... the size of the table. > What do you think? I think what you suggest should work for the simple case, but I don't think it scales well. It depends on what your needs are. Cheers, -w -- William Waites ww [EMAIL PROTECTED] magicphone.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PhoneCALL v2.6.1 - Released
Michiel van Baak wrote: On 11:21, Tue 16 Aug 05, Dustin Wildes wrote: Thanks Mark! You're right - this version is intended for the 'advanced' admin, one who is very knowledgable with Asterisk, but we are working on simplifying the interface in the next revisions that will make administration easier for most user types. Basically - think of it like this: The developer/integrator would use the 'admin' interface as-is now to configure/program the PBX. After loading the applications and setting up the accounts/extensions - they could create a 'local admin' account that would allow an office manager to add an extension, reset voicemail passwords, view reports, etc... And a user-account that would allow average-joe's (no offense to anyone named 'Joe' :) ) to easily configure their extension, review call logs - etc.. The great thing is, a system configuration can be created, exported - and ready to be loaded onto the next server. This templating can make deployment very easy and fast for Asterisk-based servers, and make life alot easier on distributors. I have the beginnings of an Administrator manual about 60% finished. It should be posted later this week or next week. Will it be possible to allow the 'local admin' to only edit specific contexts and not all. Think of this as: 1 PBX, several companies configured on it, 'local admin's per company (context) ? That would be a great feature and convince me to stop coding what I am coding now. As of right now - not currently, but it is being worked on for the next release (v2.7). We'll love to have you on board! :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
