Re: [asterisk-users] Do Asterisk requires audio codec to be installed?
Asterisk supports a whole bunch of codecs in the regular install - ulaw, alaw, gsm,ilbc being the more popular ones. A common paid codec is g729 - avbl at digium.com -rajeev On 1/29/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, Can you please tell me whether Asterisk requires any audio or video codec to be installed separately or it supports itself? Thanking you, Preeta Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Leading 0 in PRI outbound
All We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. National Numbering Plan for Landline in India is typically 0+Area Code + phone number. If it's a local number, you just dial the number without the area code. So for instance, if you want to call a number 42121234 in Delhi (Area Code 11), from any place outside of Delhi, you'd dial 01142121234 but only 42121234 within Delhi. Because of the prefix, when dialed locally, the number appears as 042121234 (which is not a valid number as there's a 0 without an area code!) There's nothing in the dial plan that is doing it. In fact, set verbose and pri intense debug indicate that the channel that's originating the cal is Zap/g0/42121234 but somehow there's a zero that gets prefixed :( Tried changing zapata.conf to include prilocaldialplan and so on but to no avail! Any help appreciated! thanks rajeev ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center Setup on asterisk
http://astguiclient.sourceforge.net/vicidial.html - supports both inbound and outbound http://queuemetrics.com/ - excellent set of metrics to measure your agents' performance! good luck -r On Dec 17, 2007 8:14 PM, Jared Smith [EMAIL PROTECTED] wrote: On Sat, 2007-12-15 at 19:06 +0200, Dovid B wrote: http://www.h6315.com/ast_docs/Asterisk%20TFOT%20v2.pdf I'm not sure who is running this website, but I'd kindly ask them to please point people to the official download at http://www.asteriskdocs.org/ instead of being an unofficial mirror. One of the important reasons for this is so that O'Reilly can better measure how many people are downloading the free version of the book versus how many people are buying the paper copy. Thanks! -Jared Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Leading 0 in PRI outbound
Yeah: we are using pridialplan=local - am using AsteriskNOW by the way. Does it require some kind of a patch? for it to understand 'pridialplan' ? My pri intense debug shows: Calling Number (len= 4) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Number not available (67) '' ] [70 0b a1 39 37 38 39 30 39 31 30 31 31] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9789091011' ] Thanks Rajeev On Dec 19, 2007 3:47 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 18 December 2007 15:22:18 Rajeev Natarajan wrote: We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. What is pridialplan set to in zapata.conf? This value sets an extra 4 bits in the PRI dialog between you and the telco. And typically, if you have it set to something like 'national', the telco will tell you you have numbers prefixed, even when you don't, because their switch software is written to make the translation. So what most people do (and what works most often) is to set pridialplan to 'unknown', which sets the bit field to all zeros and the number isn't prefixed at all at the telco switch, but simply routed based upon the number sent. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
What I expected that this will do is:
1. call the number using the string $numtodial2 - works OK
2. Set call limit to $maxcall and play a message $msgtime milliseconds
before the call - works OK
3. On connect of the call send it to the macro conn -extension s,1 with a
parameter 1002 - **does not work**
I noticed that if I interchange the L and M to read thus:
$AGI-exec('Dial',$numtodial2|30|M(conn^1002)|L($maxcall:$msgtime));
1. It dials fine
2. Transfers the call to the macro
3. ** does not** set timelimit
How can I do both - set timelimit and pass call to the Macro - is there
something that is mutually exclusive about the two functions that it does
not let me do this?
thanks
rajeev
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[asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
What I expected that this will do is:
1. call the number using the string $numtodial2 - works OK
2. Set call limit to $maxcall and play a message $msgtime milliseconds
before the call - works OK
3. On connect of the call send it to the macro conn -extension s,1 with a
parameter 1002 - Doesn't work
I noticed that if I interchange the L and M to read thus:
$AGI-exec('Dial',$numtodial2|30|M(conn^1002)|L($maxcall:$msgtime));
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Re: [asterisk-users] Problem: Using timelimit (L) and Macro (M) in Dial from AGI
Great! thanks
On Dec 3, 2007 8:31 PM, Mark Michelson [EMAIL PROTECTED] wrote:
Rajeev Natarajan wrote:
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
The problem is that you have one too many pipes ('|') in your Dial string.
Change it to this:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)M(conn^1002));
and it should work. Notice that the pipe between the L and M options has
been
removed.
Mark Michelson
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Re: [asterisk-users] Re: G729 'disappears' randomly
That's what it was... I should have posted :-) playing with /etc/mactab and nameif to fix it. -r On 4/7/07, Nikolai Lusan [EMAIL PROTECTED] wrote: On Fri, 2007-03-23 at 03:11 +0530, Rajeev Natarajan wrote: It happened again this evening and when I checked the host-id in /var/log/asterisk/messages the time when it did not register, it showed a host-id Mar 22 18:14:48 VERBOSE[2586] logger.c: == G.729 Host-ID: 90:23:3a:b7:dc:46:88:fc:cf:bb:78:a2:b8:00:75:97:34:xx:xx:xx (removing the last 6 for security) and it did not load the g729 Mar 22 18:43:18 VERBOSE[2580] logger.c: == G.729 Host-ID: 05:e5:4b:6c:0d:8b:66:fd:7a:b5:8e:a6:23:73:0b:b1:66:xx:xx:xx WORKS perfectly Any clues on why the host-id changes? IDEA: I also notice that sometimes eth0,eth1 and eth2 (Yes: i have three network interfaces) interchange on reboot. Are they related? Quite possibly, the registration program for that codec will bind to eth0 and use it as the host ID, if you change ethernetcards or re-number interfaces you will need to re-register the codec. As for the re-ordering of your network cards I would suggest you look into running udev with some rules to keep the order of the cards consistent over reboots. -- Nikolai Lusan # # # Weblog: http://lusan.id.au/~nikolai/blog # Website:http://lusan.id.au/~nikolai # # ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss
Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. Rajeev On 3/23/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk n-way call problem
Any sip debug you may have?
You might want to check your timing source. if you don't have a digium card,
to see if you have ztdummy installed correctly. Meetme requires a timing
source.
rajeev
On 3/15/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi,
i am using the n-way-call dialplan solution found on voip-info. i have
added its entry in applicationmap of features.conf file. the problem
is..its not working. to activate the n-way call i dial *0 but nothing
happens. i have played around with dtmf and codec settings but no success.
the extensions and sip configuration is below if you want to have a look. I
dont have any clue why its not working.
###extensions.conf###
[local]
exten = _XX,1,Set(DYNAMIC_FEATURES=nway-start)
exten = _XX,2,SIPDtmfMode(inband)
exten= 10,3,Dial(SIP/saad,,tT)
exten= 10,n,Hangup
exten= 11,3,Dial(SIP/riz,,tT)
exten= 11,n,Hangup
exten= 12,3,Dial(SIP/rehmat,,tT)
exten= 12,n,Hangup
[dynamic-nway]
exten = _XXX,1,Answer
exten = _XXX,n,Set(CONFNO=${EXTEN})
exten = _XXX,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)
exten = _XXX,n,Set(DYNAMIC_FEATURES=)
exten = _XXX,n,MeetMe(${CONFNO},pdMX)
exten = _XXX,n,Hangup
[dynamic-nway-invite]
exten = 0,1,Read(DEST,dial,,i)
exten = 0,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)
exten = 0,n,Dial(Local/[EMAIL PROTECTED],,g)
exten = 0,n,Set(DYNAMIC_FEATURES=)
exten = 0,n,Goto(dynamic-nway,${CONFNO},1)
exten = i,1,Goto(dynamic-nway,${CONFNO},1)
[dynamic-nway-dest]
exten = _XXX,1,Dial(SIP/${EXTEN})
[macro-nway-start]
exten = s,1,Set(CONFNO=${FindFreeConf()})
;exten = s,n,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1)
exten = s,n,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1)
exten = s,n,Read(DEST,dial,,i)
exten = s,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)
exten = s,n,Dial(Local/[EMAIL PROTECTED] ,,g)
exten = s,n,Set(DYNAMIC_FEATURES=)
exten = s,n,Goto(dynamic-nway,${CONFNO},1)
[macro-nway-ok]
exten = s,1,ChannelRedirect(${BRIDGEPEER},dynamic-nway,${CONFNO},1)
[macro-nway-notok]
exten = s,1,SoftHangup(${BRIDGEPEER})
#sip.conf###
[saad]
userid=saad
secret=1234
host=dynamic
type=friend
context=local
qualify=4000
insecure=invite,port
dtmfmode = inband
disallow = all
allow=ulaw
[riz]
userid=riz
secret=1234
host=dynamic
type=friend
context=local
qualify=4000
dtmfmode = inband
disallow = all
allow=ulaw
[rehmat]
userid=rehmat
secret=1234
host=dynamic
type=friend
context=local
qualify=4000
insecure=invite,port
dtmfmode = inband
disallow = all
allow=ulaw
#features.conf###
[applicationmap]
nway-start = *0,self,caller,Macro,nway-start
nway-inv = **,self,caller,Macro,nway-ok
nway-noinv = *#,self,caller,Macro,nway-notok
;nway-start = *0,caller,Macro,nway-start
;nway-inv = **,caller,Macro,nway-ok
;nway-noinv = *#,caller,Macro,nway-notok
--
Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] Re: G729 'disappears' randomly
More info: It happened again this evening and when I checked the host-id in /var/log/asterisk/messages the time when it did not register, it showed a host-id Mar 22 18:14:48 VERBOSE[2586] logger.c: == G.729 Host-ID: 90:23:3a:b7:dc:46:88:fc:cf:bb:78:a2:b8:00:75:97:34:xx:xx:xx (removing the last 6 for security) and it did not load the g729 So did a restart and voila! Mar 22 18:43:18 VERBOSE[2580] logger.c: == G.729 Host-ID: 05:e5:4b:6c:0d:8b:66:fd:7a:b5:8e:a6:23:73:0b:b1:66:xx:xx:xx WORKS perfectly Any clues on why the host-id changes? IDEA: I also notice that sometimes eth0,eth1 and eth2 (Yes: i have three network interfaces) interchange on reboot. Are they related? thanks rajeev On 3/22/07, Rajeev Natarajan [EMAIL PROTECTED] wrote: All, I have around 10 opteron 165 servers all running Fedora Core 5 and Asterisk 1.2.x (mostly Asterisk 1.2.16) with 15-25 channels of g729 each. They register without any problem but I had to use the codec_g729.so corresponding to the i386 version in all of them (asterisk would not start if i tried the opteron specific one). The problem: In one of the servers, we seem to lose the registration after a restart - 'show g729' simply does not work! A restart (or two!) of the server after that and things seem fine and show g729 shows the correct number of channels registered. This happens fairly randomly and have no clue why this happens only on this one machine. What we've tried: 1. permissions on codec_g729a.so - they seem fine 2. overwriting the .so file and restarting asterisk - doesn't work 3. restarting asterisk a few times - doesn't work 4. permissions on .lic file - they seem fine but none of the above seem to work. The only resolution seems to be to keep our fingers crossed while we restart the server! Ideas / thoughts more than welcome! thanks rajeev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 'disappears' randomly
All, I have around 10 opteron 165 servers all running Fedora Core 5 and Asterisk 1.2.x (mostly Asterisk 1.2.16) with 15-25 channels of g729 each. They register without any problem but I had to use the codec_g729.so corresponding to the i386 version in all of them (asterisk would not start if i tried the opteron specific one). The problem: In one of the servers, we seem to lose the registration after a restart - 'show g729' simply does not work! A restart (or two!) of the server after that and things seem fine and show g729 shows the correct number of channels registered. This happens fairly randomly and have no clue why this happens only on this one machine. What we've tried: 1. permissions on codec_g729a.so - they seem fine 2. overwriting the .so file and restarting asterisk - doesn't work 3. restarting asterisk a few times - doesn't work 4. permissions on .lic file - they seem fine but none of the above seem to work. The only resolution seems to be to keep our fingers crossed while we restart the server! Ideas / thoughts more than welcome! thanks rajeev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Warning LSP Low
All, Am running asterisk on an Opteron 165 with 4GB RAM and 1x80GB and 1x320GB SATA for a call center application (running VICIDIAL). Asterisk CLI (accessed by screen logging asterisk on startup and entering the allocated screen) gives me 'Warning LSP Low' and the voice quality goes down when this message pops up! That is, to start, we use: `/usr/bin/screen -L -d -m -S asterisk /usr/sbin/asterisk -vgc`; and then screen -r gives (among other processes) 2640.asterisk (Detached) and screen -r 2640 gives: Warning LSP Low Asterisk version 1.2.16 Zaptel 1.2.14 Wildcard TDM400 any insights into this greatly appreciated! Thanks Rajeev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Warning LSP Low
Did some more googling and grep-ping and I found that this message most likely comes from codec_g729a.so. Has anybody seen this before? Anything that we should be concerned about? Thanks rajeev On 3/16/07, Rajeev Natarajan [EMAIL PROTECTED] wrote: All, Am running asterisk on an Opteron 165 with 4GB RAM and 1x80GB and 1x320GB SATA for a call center application (running VICIDIAL). Asterisk CLI (accessed by screen logging asterisk on startup and entering the allocated screen) gives me 'Warning LSP Low' and the voice quality goes down when this message pops up! That is, to start, we use: `/usr/bin/screen -L -d -m -S asterisk /usr/sbin/asterisk -vgc`; and then screen -r gives (among other processes) 2640.asterisk (Detached) and screen -r 2640 gives: Warning LSP Low Asterisk version 1.2.16 Zaptel 1.2.14 Wildcard TDM400 any insights into this greatly appreciated! Thanks Rajeev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H extension don't work with parked calls
have you tried looking at the CLI to double check on the call flow? do make
sure that you 'set verbose 10' or something like that.
On 2/24/07, Jonathan Solano [EMAIL PROTECTED] wrote:
Hi all, I'm having a problem, with the h extension.
I have an application, when I call it check for the line requested and
then direct the call to a predefined context.
In this context I play a message (the message according to the line
called) and then park the call.
The dialplan does some other things, but my problem is that if I hung the
phone the h extension don't run, this is my dial plan
office]
include = check_voicemail
include = parking_lot
include = record_msgs
exten = fax,1,macro(RecibirFax)
exten = h,1,DeadAGI(end_logger.agi)
exten = s,1,answer()
;; pregunte por el caller id
exten = s,2,GotoIf($[${CALLERID(num)}]?4:3)
;; si no lo tiene entonces que lo cambie por 'Numero Privado'
exten = s,3,Set(CALLERID(all)=Numero Privado)
exten = s,n,SET(ARG1='2')
exten = s,n,AGI(logger.agi)
exten = s,n,hangup()
exten = ACC-4,1,playback(${SOUNDS}welcome-4)
exten = ACC-4,n,park(704)
exten = ACC-4,n,hangup
But the h extension is never called?
ideas?
--
==
Jonathan S.
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Re: [asterisk-users] Sending Email From the dialplan
I use mime-construct along with the System command - works great.
On 2/26/07, Dovid B [EMAIL PROTECTED] wrote:
- Original Message -
From: Al Bochter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 26, 2007 4:20 AM
Subject: [asterisk-users] Sending Email From the dialplan
I have looked around with no luck.
Does anyone know of a way to send an email from the dialplan.
The system that I am working on has none thing to do with VoiceMail.
This is something like the SMS command but using sending email
I am working on a prepaid alarm dispatch program for Asterisk if anyone
has any input please let me know.
I will be more than happy to write the code as Open Source for others to
use code. With help from the list.
Also I forgot to mention that you can use variables like:
exten = 1234,2,System(echo ${CALLERID} | mail -s Caller ID Info
[EMAIL PROTECTED])
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Re: [asterisk-users] Asterisk Inbound Problem
Am working with Arun on this project - here's a longer description of the problem: We've been fighting with our service provider on this issue - we seem to be getting a BYE just after we receive an ACK. They claim that it is an asterisk issue! The service provider provides only IP based authentication for inbound. We have used username-password based authentication with the same setup with *no problems* whatsoever! If we configure an Audiocodes MEdia gateway to receive the calls, there is no issue - so there's something that asterisk is doing? or asterisk-Provider gateway combo? In our efforts to mask IP, I have used PROVIDER-IP for the IP of my service provider (host) and AsteriskIP to indicate my asterisk server sip.conf [PROVIDER] type=peer disallow=all allow=g729 context=default host= fromuser=y.y.y.y port=5060 insecure=very canreinvite=no nat=yes qualify=yes CLI output: -- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack We're at 124.7.195.102 port 47698 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:8009422419@'AsteriskIP' Content-Type: application/sdp Content-Length: 183 v=0 o=root 2172 2172 IN IP4 AsteriskIP s=session c=IN IP4 AsteriskIP t=0 0 m=audio 47698 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- -- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack -- Playing 'park' (language 'en') AstSQL*CLI -- SIP read from PROVIDER-IP:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Via: SIP/2.0/UDP 221.135.102.100:5060 ;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422 Content-Length: 0 --- (9 headers 0 lines) --- AstSQL*CLI -- SIP read from PROVIDER-IP:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Via: SIP/2.0/UDP 221.135.102.100:5060 ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f Content-Length: 0 --- (9 headers 0 lines) --- Sending to PROVIDER-IP : 5060 (NAT) Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The following is an ngrep of the traffic for an inbound call - 'U' marks the begin of the packet grabbed. U PROVIDER-IP:5060 - AsteriskIP:5060 INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards: 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: sip:[EMAIL PROTECTED]:5060..From: sip:PROVIDER-IP;tag=3380960452-790279..Co ntact: sip:PROVIDER-IP:5060..Remote-Party-Id: sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID: [EMAIL PROTECTED]: 1 INVITE..Via: SIP/2.0/UDP 221. 135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events: telephone-event..Content-T ype: application/sdp..Content-Length: 206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19 CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000.. # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED] 11.2:5060..Call-ID: [EMAIL PROTECTED]: 1 INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4 ;received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED]:5060;tag=as78bcde29..Call-ID: [EMAIL PROTECTED]: 1 INVITE. .User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:800942@AsteriskIP..Content-Length: 0 # U
Re: [asterisk-users] International dialplans for Asterisk?
I think the + convention started off because different countries have different international access codes. Well, on GSM networks, + can be a part of the number to represent the international access code ( the traditional access code in India is 00 for international). So to call Digium, from my GSM phone, I can use 0018775468963 or +18775468963 and Allison will answer :) Rajeev On 12/22/06, Doug Crompton [EMAIL PROTECTED] wrote: Question... What is the purpose of the + before the number? Does anyone actually have to enter it? If so how would you do it? It is not used in the US but do I see it come in on SIP lines CID. I assume the CID ignores it in the number as I do not see it on the display. It is however stored in asterisk and when doing CID comparisions it can be a problem. Doug On Fri, 22 Dec 2006, Michiel van Baak wrote: The above number looks like: +31318787243 Try to get that from your telco, it makes life way more easy. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
Or you can look at PHP-AGI; use the php to query mysql (probably more scalable than dialplan MYSQL) Take a look at http://www.jivesoftware.org/ - perhaps some way you can use that? rajeevOn 11/10/06, Andrea Spadaccini [EMAIL PROTECTED] wrote: Ciao Ondrej, That's why I was more thinking about mysql - it is already running on my * box and remote access is no problem. Question is, if I could do the same trick you did with Asterisk DB with Mysql.Of course you can. In asterisk-addons there's the app MYSQL(), that does exactly what you want.See http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL for moredetails.HTH,--Andrea Spadaccini Multimedia Technologies Institute s.r.l.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxee lag problems ?
Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two...On 11/10/06, Jonathan Borden [EMAIL PROTECTED] wrote:I have noticed it too and do not use them anymore.. Jon-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of IraSent: Thursday, November 09, 2006 11:43 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Voxee lag problems ?At 08:48 AM 11/9/2006, you wrote: Anyone having problems with voxee since last few days or is it justme ? In peek hours i get LAGGED when i do a iax2 show peers or even1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or evenLAGGED( also shows high in peak time even when no high latency ).No problems with any other provider . Anyone else having same problem ? So it's not only me!Ira___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring 2 Asterisk servers with a SIP trunk
Asterisk B: Create an extension (just as you would if you want to connect a SIP client)Asterisk A: Have this guy register using the extension (just as you would using a SIP client like SJPhone) - You will probably have to use type=peer though. Make sure you take care of NAT and stuff like that if neededrajeevOn 10/28/06, Alok Mohapatra [EMAIL PROTECTED] wrote: Hi All, Please let me know the how to configure a SIP trunk of a asterisk Server with another one (not IAX2). Asterisk-A should register a SIP trunk with Asterisk-B server . With Regards Alok Ranjan Mohapatra Software Engineer +91 9866269992 PrimeSoft IP Solutions (P) Ltd # 917- 922,East Wing, 9th floor Block III, White House,Begumpet Hyderabad - 500016, INDIA Ph - 91-40-23418239/40 www.primesoftindia.com ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get the agent id in the recording filename
Just a wild idea:Store the filename in a variable before the call enters the queue - say RECFILENAME - and then once you know which agent has taken the call, execute an mv operation (using the system command) something like
system(mv ${RECFILENAME} ${RECFILENAME}-${AGENTNAME})i don't remember the exact syntax but something like this should work. rajeevOn 10/19/06,
David Gagnon [EMAIL PROTECTED] wrote:
Hi,
I'm sure
some else has been facing this problem. I want to record all the call coming in
my queue. I want this format: MMDD-HHMMSS-AgentID-CallerId - UniqueID. I'm
using the monitor feature inside the queue.conf. I can't use the
agents.conf monitor features because I'm using dynamic agent
(addqueuemember)
The problem I'm facing is that I
can change the filename before the call enters the queue but at this step, I
don't know which agent will get the call.
Curent dialplan :
exten =
s,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID})
exten =
s,n,Playback(recording)
exten = s,n,Queue(myJavaClub,t,,,300)
Anyone could help?
David
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Re: [asterisk-users] Anybody using inphonex service?
Tried them for all three - a tad pricey but good service imho.On 10/12/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi,I want to register with http://www.inphonex.com VoIP provider. I want to configure my Trixbox and Asterisk servers with inphonex. Anybody using this service? Mainly, I want to do three tasks. They are1) Able to make calls to USA 2) Able to make international dialing3) Able to receive incoming calls through my DID. (Are they offering DID numbers?)If anybody using this inphonex service, please tell me your feedback. Looking forward to your response. Thank you. Regards,Chandra. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2ยข/min or less. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Call centre module on Asterisk
try http://astguiclient.sourceforge.netOn 10/7/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Yes, you can easily use asterisk for a call center, start looking here http://www.voip-info.org/wiki/view/Asterisk+call+queues M Imed Imed wrote: Hi, I'm a novice in asterisk. I'm just want to know if we can develop a Call centre application on an asterisk ? ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions.On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:omar parihuana wrote: Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? )Yup. :P Thanks in advanced..--Mojo [EMAIL PROTECTED]Office Manager, Horan Company, LLC(907) 747- x112___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
http://www.electronicscience.com/ has a good IAX2 softphone called ESC SoftphoneOn 8/16/06, David Thomas [EMAIL PROTECTED] wrote:Sorry, poor reply. Yes I use it on WM5, and have not seen any problems. I admit I don'tuse it a lot, but it does seem to work fine.regards,DaveOn 8/15/06, David Thomas [EMAIL PROTECTED] wrote: Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, ChristianOn 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.htmlThe latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, DaveOn 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gui
try www.trixbox.orgasterisk source does not come with any GUIOn 8/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.All science is either physics or stamp collecting.-- Ernest Rutherford___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gui
true - i was meaning to say that it has a gui 'bundled' with it... (not to mention phpmyadmin, AGI to connect to high-level application development tools such as PHP and Perl, integrated voicemail and fax-to-email support, contact management, calling card billing and management software. autoconfiguration for Digium and Cisco phone hardware, an integrated text-to-speech system) :)mea culparajeevOn 8/1/06, Alex Robar [EMAIL PROTECTED] wrote: Trixbox is not a GUI, it's a package that includes the OS, Asterisk, a GUI, etc. FreePBX is the GUI included in Trixbox.AlexOn 8/1/06, Rajeev Natarajan [EMAIL PROTECTED] wrote: try www.trixbox.orgasterisk source does not come with any GUI On 8/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else.With warm regards.Vivek J. Joshi. [EMAIL PROTECTED]Trikon electronics Pvt. Ltd.All science is either physics or stamp collecting.-- Ernest Rutherford___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AGI cmd Record
So, this is just a wild idea. you want to send all incoming calls to a record prompt. you are probably doing something like[incoming-context]exten = s,1,SetVar(RECFILENAME=)exten = s,2,Record(${RECFILENAME})
what if you did:[incoming-context]exten = s,1,Dial(SIP/2001,120,r,L(totalrectime,warningtime))exten = 2001,1,SetVar(RECFILENAME=)exten = 2001,2,Record (${RECFILENAME})I'm guessing that the dial command with the L option will allow you to play the beep using LIMIT_WARNING_FILE variable.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial for more details on the Dial commandthis may be totally off and haven't tried it myself but probably worth a shot!
rajeevOn 7/29/06, Alexander Lopez [EMAIL PROTECTED] wrote:
There currently exist no such option. But
you are free to try to add it.
SNIP
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Re: [asterisk-users] Macro call uniqueid
Could you paste your dial plan please or email me off list? The variable space is unique for each channel - are you initiating another call within the macro? you can set the variable with a SetVar(_VARNAME=xxx) to ask the variable to be inherited in the sub-channels created from the initial one rajeev[EMAIL PROTECTED]On 7/19/06, Don [EMAIL PROTECTED] wrote: Anyone know a reason why when you jump to a macro from the dial command the uniqueid of the call changes? Or what a workaround to that would be? I have tried gettinbg the uniqueid from my AGI script while in the macro (it is different), passing the uniqueid to the macro as an ARG...tried setting my own channel variable to hold it...but it is destroyed before I can retrieve it at call end...etc..etc... ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Integrate asterisk with Database
Vidura,you would want to use some kind of IVR + php-agi to do the database operations (of course there are 10 other combinations - like Ruby - on -rails and RAGI). Quick suggestion: if you've played with asterisk before, I recommend that you look at voip-info.org for php-agi links and snapvine.com if you want to use Ruby/RAGIif you would like professional help, i suggest you post it on asterisk-biz list or contact me off-list rajeevOn 7/3/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Marcin Lukasik wrote: Have you even _tried_ to create your dialplan?And to make it worse, he copied this drivel to the Developers lists.--Chris Mason(264) 497-5670 Fax: (264) 497-8463 Int:(305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271Cell: 264-235-5670Yahoo IM: [EMAIL PROTECTED]--This message has been scanned for viruses and dangerous content by MailScanner, and isbelieved to be clean.--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Email notification
Sure... I would do the following 1. Set qualify =yesBash Script (in a cron) that doesa. asterisk -rx sip show peers foob. grep UNREACHABLE foo | wc -l mime-construct if output of the grep 1 hope this helpsrajeevOn 6/26/06, Roger Workman [EMAIL PROTECTED] wrote: Is there a way to get asterisk to send you a email when it looses or an extension doesn't re-registerRoger WorkmanBusiness DevelopmentUpperclassman/Universal Holdings LLCVoice: 304.324.3800 Fax: 304.324.3801ICQ: 4447584Website: http://www.upperclassman.netBilling Questions: billing at upperclassman.netRental Questions: rentals at upperclassman.netMaintenance: help at upperclassman.netThis e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Eric ManxPower WielingSent: Monday, June 26, 2006 1:10 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY Yes.It does not seem to cause any problems.Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95 :5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version= 1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri= sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address DEFANGED_uri=sip:[EMAIL PROTECTED] ;user=ip DEFANGED_priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED] ;tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip
http://www.voip-info.org/wiki/view/Asterisk+phonesScroll down and you will see a list of Softphones that you can choose from. best way to test it, imho, use: 1. Echo test - http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Echo2. configure another sip phone on another PC and call! you may also asterisk -r on the asterisk server to get to asterisk CLI and see call progress as you make calls. rajeevOn 6/8/06, issam [EMAIL PROTECTED] wrote: hello how can i configure asterisk to use soft sip phone and when asterisk is running how can I know he work correctly thanks ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-1001 behind NAT - Internet Asterisk box --BOUNTY!
This worked for me yesterday: -Please replace your actual extension number where it says extensionnumber and password in passwordOn asterisk:[extensionnumber] username=extensionnumbertype=friendsecret=passwordrecord_out=Adhocrecord_in=Adhocqualify=yesport=5060nat=yesmailbox=extensionnumber@devicehost=dynamicdtmfmode=rfc2833 context=from-internalcanreinvite=noConfiguring the SPA 1001http://ip_of_sipura/admin/advancedClick on Line 1NAT Keep alive Enable = yesProxy=ip_address_of_asterisk user id: extensionnumberpassword:passwordSubmit all changes--Didn't have to do port forwarding on the NAT router or anything.hope this helpsRajeev On 5/19/06, Freddy Setiawan [EMAIL PROTECTED] wrote: Well, just make sure the sip.conf for your extension has the nat=yes, and don't forget to open the firewall hole for the service to run. Best Regards, Freddy Setiawan Senior Programmer Simpleware Solution [EMAIL PROTECTED] Yahoo Messenger: [EMAIL PROTECTED] Msn Messenger: [EMAIL PROTECTED] Skype: FreddySetiawan Mobile Phone: +65-98279537 :: Simple is Everything, Nothing is Complex :: From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Lachek Butalek Sent: Friday, May 19, 2006 3:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-1001 behind NAT - Internet Asterisk box --BOUNTY! Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. I'm using a GNet VP168I (same as the PA168V) and it works fine even behind a NAT which is itself behind a corporate firewall... Just a thought. On 5/17/06, Eric Lyons [EMAIL PROTECTED] wrote: I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet.It works fine to my local [EMAIL PROTECTED] box. I've tried... many things. I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working.Any Sipura experts out there? Eric. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] virtual extension per user ?
If the users have a bluetooth device like a cellphone-with-bluetooth or their laptop, this might work: http://mundy.org/blog/index.php?p=78 - you'll have to modify the script in the tutorial a bit. essentially - you have a presence server at the two offices - when they enter the building, the bluetooth device registers with the presence server and the corresponding phone comes alive. works great for us (as long as the bloke doesn't leave the cellphone at home) rajeev -- Chief Technology Officer Gyantec Consulting (I) Pvt. Ltd. Chennai, INDIA Phone: +91-44-4205-4446 Mob : +91-944-407-2925 Fax : +91-44-4205-4546 VoIP : +1-360-519-5969 Alex Ongena wrote: Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the single extension XXX. I know you can 'forward' calls to other extensions, but when people go from Office 1 to Office 2, they forget to enable their forward in Office 1 to Office 2. I like a solution where they can say 'Please register me, I'am now sitting in Office 2'. The moment after 'registration', when you call XXX, the phone in Office 2 will ring. In all places I use Asterisk 1.2.1 with bristuff, Cisco 7940/60 phones with Sip and some Sip softphones. Any hints or tricks to get this behaviour ? Thanks Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Individual SIP account how to make it Trunk
Have you tried using the Trunk Sequence AMP -- Setup -- Outbound Routing Seems to work for us! Rajeev -- Chief Technology Officer Gyantec Consulting (I) Pvt. Ltd. Chennai, INDIA Phone: +91-44-4205-4446 Mob : +91-944-407-2925 Fax : +91-44-4205-4546 VoIP : +1-360-519-5969 Dovid Bender wrote: Not sure if this will help but for multiple reasons we send all calls thru astcc. In astcc you specify what route you want it to use. If a route isnt available then it tries the next one you specified and so on. --- Jolly M. Recto [EMAIL PROTECTED] wrote: Hi, i have diffirent provider example(3 single account in deltathree, 4 account in packet8 and so on) . How this possible to make the three individual sip account in deltathree act as trunk so that i cannot get a busy call. If line one fail goto line 2 then line 3 or another trunk line 1 then line 2 then line3I read it in asterisk at home but the script i am copying is not working . any help is very much appreciated.. TIA //jollyr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] regarding connecting to AMP
http://mundy.org/blog/index.php?p=93 http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki (Chapter 4 and 7) The above links have some excellent documentation. www.voip-info.org specifically has some really good setup examples. Recommend you go through those... -R Sohail Arham wrote: hi alli have intalled [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] successfully and now the problem is that how can i connect to AMP so that i would be able to configure it.actually i have following setup... one [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] machine and two other machines i want that these two clients machine can be able to call each other through using [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] box.i connect this [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] box to the hub...(simple hub) ...now tell me what ip scheme i would use to configure it ...and how it would be possible to complete my task...one more thing i have also xlite sip phone ...i will call these two machine through these sip soft phonesnow plz temme complete idea becaz i have no good experience about it.i shall be thankful to you BYE -- Muhammad Sohail Arham U.E.T. Lahore Phone No. 0321-4422406 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AAH out bound routing problem
if you are using AAH, please post extensions.conf, extensions_additional.conf - also send us more info on your phones. thanks rajeev ram wrote: Hi all of them thanks for the quick reply i was tried adding 9 as well as 00 but i get number invalid if i put any of the digits what kind of config files need to post here to resolve the problem please assists ram On 1/27/06, *Michael Collins* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ram, On my AAH the stock dial plan requires a 9 first. For kicks, try dialing 919197543700 and see what you get. -MC *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] *On Behalf Of *ram *Sent:* Friday, January 27, 2006 6:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] AAH out bound routing problem Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
