Re: [asterisk-users] Asterisk CDR Problem
Hi Max, In CDR configuration for Pstgres Database, I have copied cdr_pgsql.so module file from other Asterisk System. May be that happen in this my problem I have Installed Asterisk 1.4 on CentOS-5 (64bit). So, exactly what in both system which I have made recently the file size of cdr_pgsql.so is different one is 128.2 kb and one is 130.5 kb when I copied 130.5 file in /usr/lib/asterisk/modules and cdr_pgsql.so can load which is I describe below localhost*CLI module load cdr_pgsql.so == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Loaded cdr_pgsql.so = (PostgreSQL CDR Backend) localhost*CLI -- And OUTPUT on CLI --- Below localhost*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom CDR registered backend: pgsql localhost*CLI Now data are inserting in Postgres Database. Thank you Max for your Supports. But is it possible there are two different file size ? And when I install Asterisk 1.4 in /etc/asterisk directory no one files are copied. Now that problem is solve. With Regards. Hiren Mistry Max Alex wrote: Hi, let me know that you have configured properly in res_pgsql.conf in asterisk with proper, and it is connected properly to database with database details. Thanks, Max Alex Voip Developer On Fri, Aug 29, 2008 at 10:26 AM, Hiren Mistry [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi , I have check zapte.conf in and after make some correction that problem solve. But now I am facing other problem. We are using here Postgres Database and the data from CLI it can't insert in Postgres Database. I have also here mention below cdr_pgsql.conf, modules.conf and cdr.conf cdr.conf -- Below [general] [csv] usegmtime=yes ;log date/time in GMT loguniqueid=yes ;log uniqueid loguserfield=yes ;log user field -- cdr_pgsql.conf -- Below [global] hostname=localhost ;hostname=122.160.10.81 http://122.160.10.81 port=5432 dbname=asterisk password=postgres user=postgres table=cdr -- modules.conf -- Below [modules] autoload=yes ;preload = res_odbc.so ;preload = res_config_odbc.so noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so noload = pbx_kdeconsole.so load = res_musiconhold.so noload = chan_alsa.so -- OUTPUT on CLI --- Below localhost*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom localhost*CLI also when I load manually cdr_pgsql.so on CLI then it show error which is also I describe below localhost*CLI module load cdr_pgsql.so [Aug 29 10:22:21] WARNING[8984]: loader.c:362 load_dynamic_module: Error loading module 'cdr_pgsql.so': libpq.so.5: cannot open shared object file: No such file or directory [Aug 29 10:22:21] WARNING[8984]: loader.c:614 load_resource: Module 'cdr_pgsql.so' could not be loaded. So, Please guide me for load Postgres module in asterisk for CDR Database. Subject: Re: [asterisk-users] Asterisk CLI Show Error :- (**Unknown**) instead of (Zap/22-1, ) From: Max Alex [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Date: Thu, 28 Aug 2008 11:48:16 +0530 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hi Hiren, Have you properly configured the zap channels in asterisk, which device have you configured in asterisk with zaptel? let me know the dial plan for ivr. Thanks, Max Alex Voip Developer On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Everybody, I am planning to make a new IVR on Asterisk I have Installed zaptel , libpri, asterisk, asterisk-addon on CentOS 5 I also start service of zaptel and asterisk it start successfully. But when goto asterisk CLI prompt and check this IVR then all call string with (**Unknown**) instead of (Zap/22-1, ) and I have also 3
[asterisk-users] Incoming Calls via SIP Trunks
Hello, i have one question regarding incoming SIP INVITES. I have a testbed where i have 5 extnsions : 6001 - 6005 Domain : domainA.com Then i have configured a sip trunk, where my PBX registers to a foreign SIP Proxy. All is working fine, until following scenario: Incoming call from [EMAIL PROTECTED] (SRV exists,user also exists in pbx as extension, but different domain!) When i try this, the pbx answers with an proxy-auth. When i remove extension 6002, all is working again as aspected. Question: does asterisk not verify the domainpart ? How are incoming INVITES processed ?! I´m using the latest SVN Branch 1.4 br, Andreas M. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold is not working
I have made class for MOH uploaded a mp3 file to the folder. Now I am using this class for music on hold during dialing. Now when call has been established, I put the other end on hold. So from that end I should listen uploaded file. But I am not getting audio. From memory, you need to install asterisk-addons in order to play mp3 file. The default audio file is .gsm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
Have you looked at PRI-BRI Fail-over-Switches ? 2008/8/27 Jeremy Mann [EMAIL PROTECTED] We've done the asterisk passthrough route, but if the asterisk box is down for whatever reason both systems are down. Splitter wasn't the right word, but yes I see your point, I'll look into the Adtran. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, August 27, 2008 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Splitter Jeremy Mann wrote: I know I could probably achieve the same thing with a 3 port PRI card in a server, but I'd like something braindead easy to configure from both a hardware and software perspective. Anything you use is going to (essentially) be a 3-port ISDN PRI capable switch, because that is the only way to accomplish what you need. There really isn't any way to 'split' a PRI, unlike a T1 using CAS signaling which can be 'split' using a drop-and-insert multiplexer. If you don't want to use a small PC with a 3-port T1 card in it, you can use something like an Adtran Atlas to do the job. Alternatively, just use a 2-port T1 card in the Asterisk server, and run the PRI *through* the Asterisk server on the way to the other PBX. That's the most common way to do what you want to do. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
Doug Lytle wrote: This is what we do. Along with an ADIT 600 (eBay special) Just as a side note, I got very lucky on eBay and got my Adit and FXO 8 card for $50 each. I was tickled pink! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
Anthony Francis [EMAIL PROTECTED] writes: Lets not forget that the DECT specification does allow for data transmission. THere is no reason that in the future you would not be able to have integrated services over DECT. The DECT data rate is way too low for integrated services. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
The DECT data rate is way too low for integrated services. Does anyone know the actual data rate on DECT? I've never built any wireless mobility apps, but I would assume the bitrate required would be quite low, being mostly XML text. In comparison to the audio data stream it would seem to be the probverbial fart in a windstorm even though it would perhaps not be surprising if it were only designed to carry CNAM strings etc, corroborating Benny Amorsen post. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Heist of MagicJack SIP credentials?
While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a preferred DID number. Has anybody done, or know of someone who has done this? -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
Have you looked at PRI-BRI Fail-over-Switches ? We've done the asterisk pass-through route, but if the asterisk box is down for whatever reason both systems are down. Look at this brand new failover device: http://www.rhinoequipment.com/1portfail.html http://www.rhinoequipment.com/Single%20Port%20Failover%20Datasheet%201-22-2008.pdf I've done a fair amount of research and this one is easier, cheaper, more robust and more flexible than other solutions I've looked into. I'm getting one in a few days. I talked about it a little bit yesterday on the VoIP users conference. http://recordings.talkshoe.com/TC-22622/TS-137957.mp3 My ramble about about Redfone and Rhino is about three fifths of the way through the recording. It's a little winded, but I was speaking top-of-mind. You can the Rhino product for T1/E1/BRI or ANALOG trunks. There are several good solutions for PRI, including the redfone bridge: http://www.red-fone.com/Products/fonebridge2/ The device from Rhino is in my opinion is a HUGE no-brainer for BRI or Analog failover scenarios -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
Has anyone ever really, truly, actually held on to a Wi-SIP call while moving from the range of one AP to the range of another AP in the same network? Let's say a 'YES' only counts if you had a bona-fide handoff. In other words, you began in place 'A' (within range of AP#1 but OUTSIDE the range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but completely outside the range of AP#1) WITYOUT dropping the call. Supposedly it's possible with compliant hardware using 802.11f - Inter-Access Point Protocol (IAPP), but given how ALL standards ALWAYS work together PERFECTLY, 100% of the time :-), I'm guessing that it doesn't work. Can anyone speak to this from experience? -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
On Sat, 30 Aug 2008 11:51:49 -0500, Karl Fife wrote: Has anyone ever really, truly, actually held on to a Wi-SIP call while moving from the range of one AP to the range of another AP in the same network? Let's say a 'YES' only counts if you had a bona-fide handoff. In other words, you began in place 'A' (within range of AP#1 but OUTSIDE the range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but completely outside the range of AP#1) WITYOUT dropping the call. Supposedly it's possible with compliant hardware using 802.11f - Inter-Access Point Protocol (IAPP), but given how ALL standards ALWAYS work together PERFECTLY, 100% of the time :-), I'm guessing that it doesn't work. Can anyone speak to this from experience? -Karl Karl, I'm guessing that it was not common. 802.11f handoffs reportedly take 100ms which is considered too long for streaming applications like voice and video. The 802.11r standard was only agreed upon and released days ago. This specifies FAST BSS transition specifically to saisfy such applications. Not sure if any hardware supports this as yet. http://en.wikipedia.org/wiki/IEEE_802.11r Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
On Saturday 30 August 2008 11:51:49 am Karl Fife wrote: Let's say a 'YES' only counts if you had a bona-fide handoff. In other words, you began in place 'A' (within range of AP#1 but OUTSIDE the range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but completely outside the range of AP#1) WITYOUT dropping the call. wouldn't the ap ranges have to have *some* overlap, lest the basic network connection be dropped, whereby dropping the voip call? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
On Sat, 30 Aug 2008 09:05:58 -0500, Karl Fife wrote: The DECT data rate is way too low for integrated services. Does anyone know the actual data rate on DECT? I've never built any wireless mobility apps, but I would assume the bitrate required would be quite low, being mostly XML text. In comparison to the audio data stream it would seem to be the probverbial fart in a windstorm even though it would perhaps not be surprising if it were only designed to carry CNAM strings etc, corroborating Benny Amorsen post. -Karl Actually, back in January when I was researching DECT wrt to the snom review that I was writing I found the situation better that you expect. The very first DECT product, back in the early 90's, was a data networking device from Olympia, not voice at all. DECT was designed from the start to handle voice and data. Here are some resources: http://www.dectweb.com/Introduction/answers.htm http://en.wikipedia.org/wiki/DECT http://www.dect.org/ Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Congestion in Outgoing call through PRI
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0501125
-- Zap/1-1 is proceeding passing it to SIP/2000-081b9938
-- Zap/1-1 is making progress passing it to SIP/2000-081b9938
-- Channel 0/1, span 1 got hangup request
-- Zap/1-1 is circuit-busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/2000-081b9938, ) in new stack
== Spawn extension (default, 920501125, 2) exited non-zero on
'SIP/2000-081b9938'
Zaptel.conf
loadzone=us
defaultzone=us
#Sangoma A101 port 1 [slot:4 bus:5 span:1] wanpipe1
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Zapata.conf
-
[trunkgroups]
[channels]
context=default
usecallerid=no
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A101 port 1 [slot:4 bus:5 span:1] wanpipe1
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel =1-15,17-31
extensions.conf
---
[globals]
;CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g0 ; Trunk interface
[from-pstn]
exten = 4392839,1,Answer
exten = 4392839,2,Wait(1000)
exten = 4392839,3,Goto(default,1000,1)
[default]
exten = 1000,1,Playback(transfer)
exten = 1000,2,Hangup
exten = _92X.,1,Dial(${TRUNK}/${EXTEN:2},,)
exten = _92X.,2,Hangup
sip.conf
---
[1000]
type=friend
secret=1000
host=dynamic
disallow=all
allow=alaw
allow=ulaw
Where i m on the mistake
Shariq
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Re: [asterisk-users] Heist of MagicJack SIP credentials?
On Sat, 30 Aug 2008 11:31:26 -0500, Karl Fife wrote: While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a preferred DID number. Has anybody done, or know of someone who has done this? -Karl No, but if you did this and posted it online you get HUGE traffic! I posted only one or two items about Magic Jack over the months and they generate constant page views. I don't really undertand why. Some folks are going to what I think are considerable efforts to make the Magic Jack USB device work with thin clients so that they don't need their PC on 24/7. That's a lot of effort. If someone published a way to make a plain old ATA word with MJ you might even force them to change their operating practices. Remember that their little client gets fed adverts and they track all your calling habits in order to target those adverts. You call Domino's to order a pizza and they're likely to show you a Papa John's ad next time. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] beta9: how to set callerid on incoming iax?
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
sean
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Re: [asterisk-users] beta9: how to set callerid on incoming iax?
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
No not the channel, it's the blank before first stanza. Why's it blank?
sean
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Re: [asterisk-users] Congestion in Outgoing call through PRI
2008/8/30 Shariq Khan [EMAIL PROTECTED]
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0501125
-- Zap/1-1 is proceeding passing it to SIP/2000-081b9938
-- Zap/1-1 is making progress passing it to SIP/2000-081b9938
-- Channel 0/1, span 1 got hangup request
-- Zap/1-1 is circuit-busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/2000-081b9938, ) in new stack
== Spawn extension (default, 920501125, 2) exited non-zero on
'SIP/2000-081b9938'
Zaptel.conf
loadzone=us
defaultzone=us
#Sangoma A101 port 1 [slot:4 bus:5 span:1] wanpipe1
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Zapata.conf
-
[trunkgroups]
[channels]
context=default
usecallerid=no
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A101 port 1 [slot:4 bus:5 span:1] wanpipe1
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel =1-15,17-31
extensions.conf
---
[globals]
;CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g0 ; Trunk interface
[from-pstn]
exten = 4392839,1,Answer
exten = 4392839,2,Wait(1000)
exten = 4392839,3,Goto(default,1000,1)
[default]
exten = 1000,1,Playback(transfer)
exten = 1000,2,Hangup
exten = _92X.,1,Dial(${TRUNK}/${EXTEN:2},,)
exten = _92X.,2,Hangup
sip.conf
---
[1000]
type=friend
secret=1000
host=dynamic
disallow=all
allow=alaw
allow=ulaw
Where i m on the mistake
Shariq
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Update to latest libpri and tell us if it still demonstrates the problem,
use HEAD version.
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Re: [asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
wouldn't the ap ranges have to have *some* overlap, lest the basic network connection be dropped, whereby dropping the voip call? Indeed you're right. You'd have area covered by AP 'A' only, AP 'B' only and area of AB overlap, Picture a venn diagram: http://upload.wikimedia.org/wikipedia/commons/5/56/Venn-diagram-AB.png -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] beta9: how to set callerid on incoming iax?
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
No not the channel, it's the blank before first stanza. Why's it blank?
So I set up ten special iax-in* contexts in extensions.conf, which set
callerid and then goto [longdistance]. Seems a weird way to do it, but
it works.
sean
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Re: [asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
On Saturday 30 August 2008 01:35:10 pm Karl Fife wrote: Indeed you're right. You'd have area covered by AP 'A' only, AP 'B' only and area of AB overlap, Picture a venn diagram: http://upload.wikimedia.org/wikipedia/commons/5/56/Venn-diagram-AB.png right. it's just your inital description made it sound as if there was NO overlap. is the delay from the switchover too much to cover the call without dropping? i was thinking, on a laptop, where you are not streaming realtime, a few seconds you might not notice, but on a voip phone during a call... -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue when dialing multiple extensions using ------Please Help
Krunal Patel wrote: Hi, I have a simple dialplan. [test] exten = _X.,1,Dial(SIP/1000SIP/1002) I have registered user whose context is test. Now I am dialing any number, so it will enter into test context. It will dial 1000 1002 both. Both keeps ringing. Now the problem is, when any of them answer, another one keeps ringing for 10 to 15 sec. Please help me , what's wrong here. Thanks, Krunal Patel It's hard to say for certain based just on the details given, but it could be that the SIP CANCEL generated by Asterisk to the second phone is being rejected or is not being generated in the first place. Seeing a SIP debug from the call would help to diagnose the problem. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones (Tzafrir Cohen)
The Siemens DECT line are open source. Not broadly available in the US though. What does this mean? Could you please provide a link for more information? here's a link to the Siemens open source dect/wifi phones if that's what you are looking for. http://gigaset.siemens.com/shc/0,1935,hq_en_0_121782_rArNrNrNrN,00.html Freddi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
Michael Graves [EMAIL PROTECTED] writes: DECT was designed from the start to handle voice and data. 553kbps isn't particularly useful today. Even 3G is faster. Not that I have ever seen products offering more than ISDN speeds, and I believe that was before 2000. DECT is excellent, but it appears that the market expects it to be replaced by 802.1something once power consumption gets low enough. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers on AgentLogin()
What did you try and how did it fail? Are you using the t option in queue? On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote: So, no answers or is this thread going to remain unanswered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. -- Sent from Gmail for mobile | mobile.google.com Matt Riddell Director VentureVoIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with DTMF on IVRs
last time i had this issue with teliax, they recommended to upgrade to 1.4 On Fri, Aug 29, 2008 at 3:44 AM, Chris Mason [EMAIL PROTECTED] wrote: I tried DTMFmode=auto and it did not help. Any further ideas? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] security on localhost connections
Asterisk Users - We are presently try to operate a hybrid GSM/Asterisk cellular basestation at the Burning Man Festival in the Nevada desert. (See http://openbts.sourceforge.net). The architecture is basically one where cell phones are presented to Asterisk as SIP users, using the IMSI as the SIP user ID for convenience. (It's running off of a wind turbine is the middle of a dust storm as my alkali-abused hands type this.) When we first got this system running, we were getting hammered with service requests from phones that people left turned on. We tried sending the magic GSM codes for no roaming here, but some of them just kept coming back. It was like a denial of service attack. We figured out that the best way to shut those phones up was just to accept their registrations. We'd send a corresponding SIP registration to Asterisk, that would fail, but we'd report success to the GMS handset anyway so that it would think it had service and stop retrying the registration. Now we've discovered a new problem: Asterisk lets these non-existent make calls even though they are not listed as users in sip.conf. We suspect that is happening because they are all localhost connections, and therefore bypassing some kind of authentication check. These calls also show up in the CDR, but with the SIP ids of real, provisioned SIP users instead of the IMSIs of the phones that are actually making the calls. Any ideas how this is happening or how to fix it? -- David David A. Burgess Kestrel Signal Processing, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heist of MagicJack SIP credentials?
MagicJack has back hacked for some time http://revolution.hackthisbox.com/magicjack/readme Cory J. Andrews Director New Market Initiatives VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Saturday, August 30, 2008 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Heist of MagicJack SIP credentials? On Sat, 30 Aug 2008 11:31:26 -0500, Karl Fife wrote: While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a preferred DID number. Has anybody done, or know of someone who has done this? -Karl No, but if you did this and posted it online you get HUGE traffic! I posted only one or two items about Magic Jack over the months and they generate constant page views. I don't really undertand why. Some folks are going to what I think are considerable efforts to make the Magic Jack USB device work with thin clients so that they don't need their PC on 24/7. That's a lot of effort. If someone published a way to make a plain old ATA word with MJ you might even force them to change their operating practices. Remember that their little client gets fed adverts and they track all your calling habits in order to target those adverts. You call Domino's to order a pizza and they're likely to show you a Papa John's ad next time. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
On Sat, 30 Aug 2008 23:41:16 +0200, Benny Amorsen wrote: Michael Graves [EMAIL PROTECTED] writes: DECT was designed from the start to handle voice and data. 553kbps isn't particularly useful today. Even 3G is faster. Not that I have ever seen products offering more than ISDN speeds, and I believe that was before 2000. DECT is excellent, but it appears that the market expects it to be replaced by 802.1something once power consumption gets low enough. Clearly DECT is not a data solution in the larger sense. But Wifi has a long way to go before it can truly compete with DECT solutions available today. You have to really truly need the converged data+voice aspect of wifi for it to make sense. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] security on localhost connections
On Saturday 30 August 2008 19:15:36 David Burgess wrote: Now we've discovered a new problem: Asterisk lets these non-existent make calls even though they are not listed as users in sip.conf. We suspect that is happening because they are all localhost connections, and therefore bypassing some kind of authentication check. These calls also show up in the CDR, but with the SIP ids of real, provisioned SIP users instead of the IMSIs of the phones that are actually making the calls. Any ideas how this is happening or how to fix it? Generally, this is because your SIP users don't have passwords. Force passwords on all of your SIP devices, and alternate SIP endpoints won't be able to make calls without that corresponding user/password. The reason this happens is due to the matching sequence, where Asterisk prefers a match with no password (and where the host is dynamic) when all other searches fail to produce a match. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intermittent rejected because extension not found On Incoming DID
I have a DID with budgetphone.nl, which has worked fine for quite some time. For the last few weeks, most (but not all of the time), the incoming call does not go where it is supposed to, but instead the following message show on the console: [Aug 30 19:47:28] NOTICE[5161]: chan_sip.c:13865 handle_request_invite: Call fro m '' to extension '3120333' rejected because extension not found. (NOTE: I have substituted 333 for the last 7 digits of the DID for purposes of posting.) Here is the relevant portion of sip.conf: [3120333] type=friend username=3120333 secret= host=budgetphone.nl fromuser=3120333 fromdomain=budgetphone.nl nat=yes authuser=3120333 dtmfmode=rfc2833 context=amsterdam insecure=very canreinvite=no disallow=all allow=ulaw qualify=no port=5060 In extenstions.conf, I have the following context: [amsterdam] exten = 3120333,1,Set(CALLERID(name)=Amsterdam) exten = 3120333,2,Set(CALLERID(number)=33) exten = 3120333,3,GoTo(voicepulse-incoming,5379,1) This configuration has worked for quite some time (over a year), but only recently has stared acting up. What I don't understand is the intermitten nature - I would htink it shoudl wither work, or not work. Anyone out there have any ideas? --Robert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
