Re: [asterisk-users] Grandstream GXP2000 - copy configuration from handset
Thanks for all of your suggestions - I shall try both! Glen On 10/9/2011 14:42, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Le 09/10/2011 03:40, Silverthorne Wystead a écrit : I have a Grandstream GXP2000 and I would like to use tftp or some other utility to grab the configuration from it. Anyone have any bright ideas? gsutil works for me: http://www.pkts.ca/gsutil.shtml Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk6R6woACgkQuu7Rv+oOo/iLoQCfa22qoGXgca5yjkykbamzAzDL 8K4An1LOB8owlQdyhLAqZp5YIArsL/BM =yC6f -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
Hey Elliot;
Would you mind posting your dialplan for your Google Voice config? I am
having a hell of a time getting it to do *anything*.
Perhaps I am just fat-fingering.
Would you mind? Thanks in advance.
Glen
On 6/13/2011 19:02, Elliot Murdock wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING:iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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[asterisk-users] Question about voip.ms service.
Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms http://voip.ms/ VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start troubleshooting something like this? Thanks Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x
Hello Folks; Perhaps I am chasing my tail here. Before I go any further, is this compatible/supported in Asterisk 1.6x? If so, I would be willing to post any manager.conf or http.conf snippets needed. When I attempt to open the Asterisk Web GUI, I get a 'page not found'. I am sure this is something really minor - something silly that I missed. Any words of wisdom? Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
Works well - however, I see you included the API access. Are there more parameters that we can pass to get more information? Example, when we go to the web site, it gives you the City/State/Province/Postcode and carrier. G On 5/29/2011 07:47, Michael R. Wally wrote: FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 You can monitor the stats, including the current queue size, at freecnam.org API Results will continually improve as the database grows, so please be patient with limited results at this early stage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using MixMonitor()
Hello Folks; I appreciate all of the help so far - thanks. Another question: I am using MixMonitor() to record calls and I would like to include the called number/extension in the filename: In my dialplan, I am able to save the file with the caller id in the filename. However, what I am a little unsure about is the incoming number/called number/extension - passing that information on to part of the filename. Does anyone follow me? Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - methodology
Dan et al; Okay - I have declared DYNAMIC_FEATURES=MixMonApp in the [global] section of my extensions.conf I dial into my trunk, the softphone rings, I answer and I press '*1' - I hear the tones, but I see no indication in the Asterisk CLI and I see no .wav file being created. I must still be missing some subtle little thing. Wow, this is taking on a life of it's own. What am I missing? Not reading the DTMF tones. Thus not executing the macro. Keep in mind, that if I execute the macro manually (put in right in my extension declaration in extensions.conf, it works) Let me know if you want to see anything (parameters, etc) Thanks Glen On 4/9/2011 20:51, Dan Journo wrote: If you don't want to record every call, you can give the operator the option of press *1. We did this by adding the following to features.conf:- MixMonApp = *1,self/both,Macro,mixmon As brought up in another post, I forgot to add the following:- DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a per channel basis in extensions.conf. Thanks to Warren Selby from http://www.selbytech.com for pointing that out. Dan Journo Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - methodology
Hi Dan et al; I had actually done a sip reload, dialplan reload, module reload res_features.so and logger reload. However, upon seeing your email, I restarted the Asterisk server completely to see if I had missed anything. I still see the same behaviour. I am at a loss. Glen On 4/10/2011 14:37, Dan Journo wrote: I set the logger.conf to show reading of DTMF tones as per your instructions below. This is what I see: [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin '*' received on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF begin passthrough '*' on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end '*' received on SIP/6000-002e, duration 186 ms [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end accepted with begin '*' on SIP/6000-002e [Apr 10 11:57:19] DTMF[14783] channel.c: DTMF end passthrough '*' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin '1' received on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF begin passthrough '1' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end '1' received on SIP/6000-002e, duration 193 ms [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end accepted with begin '1' on SIP/6000-002e [Apr 10 11:57:20] DTMF[14783] channel.c: DTMF end passthrough '1' on SIP/6000-002e It looks like Asterisk hasnt added the new details from features.conf. You may need to fully restart Asterisk in order to get this to work. Dan Journo Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - methodology
Hey! I did a little bit of digging - and I solved my issue! Apparently, in my extensions.conf, I specified the wrong variable. I had DYNAMIC_FEATURES=callrec (which is the name of my macro) I changed it to DYNAMIC_FEATURES=MixMonApp, which is what is it aliased to in the features.conf. Looking back through the email trail, I think I must have overlooked that. My bad. However, I thank all of you for your patience and help. Nice to have friends in high places! Thank you again. Guinness for everyone! Glen On 4/10/2011 17:09, Dan Journo wrote: I am at a loss. Can you pastebin the following:- - Run asterisk-cvvvddd and paste the output - Pastebin your features.conf - Pastebin your extensions.conf I'll see if I can spot anything obvious. Dan Journo Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - methodology
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the MixMonApp =
*1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I
cannot get it to work.
Steps.
1. added the example macro to the dialplan in extensions.conf
2. added the line MixMonApp = *1,self/both,Macro,mixmon to the
features.conf file under [applicationmap]
3. sip reload / dialplan reload / reload res_features
4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)'
5. make incoming call - answer with SIP phone
6. I press *1 on the keypad, I hear the tones, but it does not begin
recording
7. see nothing in the CLI and no new files get created in
/var/spool/asterisk/monitor directory.
What am I missing? Probably something simple.
Any words of wisdom?
Glen
On 4/6/2011 07:29, Dan Journo wrote:
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there
that
I can try out?
We give our clients to option of either recording all calls, or
allowing the operator to press *1 during a call to start recording
manually.
Using Asterisk 1.4, this is what we do:-
We created a Macro in extensions.conf like this:-
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} =
]?startrec:donothing)
exten = s,n(startrec),GotoIf($[${ARG1}=]?beep:nobeep)
exten = s,n(beep),Playback(/var/lib/asterisk/sounds/rec1)
exten = s,n(nobeep),Set(XAD=1)
exten = s,n,MixMonitor(FILENAME.wav,b)
exten = s,n(donothing),MacroExit
(please note, FILENAME.wav is usually ${UNIQUEID:0:10}, but I changed
it to make it easier for you to understand. You'll need to change it
back to something like ${UNIQUEID:0:10}.wav if you are recording
multiple calls because otherwise they'll be constantly saved to
FILENAME.wav and you'll lose all the previous calls.)
(please note, /var/lib/asterisk/sounds/rec1 is a beep tone so that the
operator knows that he's successfully started the recording.)
Then to recording every call, we add this before the
DIAL(SIP/extension) command in extensions.conf:-
exten = _9.,14,Macro(mixmon,nobeep)
If you don't want to record every call, you can give the operator the
option of press *1. We did this by adding the following to features.conf:-
MixMonApp = *1,self/both,Macro,mixmon
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBX http://www.keshercommunications.com/hostedpbx.html
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[asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.
Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp = *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to work. Steps. 1. added the example macro to the dialplan in extensions.conf 2. added the line MixMonApp = *1,self/both,Macro,mixmon to the features.conf file under [applicationmap] 3. sip reload / dialplan reload / reload res_features 4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)' 5. make incoming call - answer with SIP phone 6. I press *1 on the keypad, I hear the tones, but it does not begin recording 7. see nothing in the CLI and no new files get created in /var/spool/asterisk/monitor directory. What am I missing? Probably something simple. Any words of wisdom? Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call recording - methodology
Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Firewalling and Asterisk
Forgive my ignorance on this as I am still fairly new to Asterisk. I have noticed lately that there have been several attempts to hack our Asterisk server. I see multiple attempts to log in with a particular extension from the same IP address, perhaps hundreds of times per second. It causes the overhead to spike to ~100%. It is more of a pain in the ass than anything. So far what I have been doing is adding a drop of this particular IP address to my iptables configuration. This makes that particular one stop and overhead drops back to normal. What I would like to know is: 1. has anyone else seen this? 2. what is the best way of prevention? We are awaiting our Cisco firewall, but I can implement a software solution in the meantime (Shorewall). So, I am wondering if anyone has a firewall/IP tables statement that keep out unauthorised users? No one seems to get in as we use really strong passwords. However, the attempts cause our Asterisk server to grind almost to a halt. I cannot even connect with a SIP phone when this happens. Any words of wisdom for me? Thanks! Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way voice with Asterisk
Let me explain: When I dial into Asterisk ( I have a SIP trunk - which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing - the extension rings on the Asterisk server (you can see it in the CLI and hear it at the computer), and the softphone rings However, when you answer the SIP softphone , you can only hear the voice FROM the softphone out. Where would I start to troubleshoot this? I am a little clueless! Thanks for all of your help. Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running Linux on 2010-06-10 14:32:34 UTC Sip Settings: Global Settings: SIP Port: 5060 Bindaddress:0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Allow subscriptions:Yes Allow overlap dialing: Yes Promsic. redir: No SIP domain support: No Call to non-local dom.: Yes URI user is phone no: No Our auth realm asterisk Realm. auth:No Always auth rejects:No Call limit peers only: No Direct RTP setup: No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events:Off IP ToS SIP: none IP ToS RTP audio: none IP ToS RTP video: none T38 fax pt UDPTL: No RFC2833 Compensation: No SIP realtime: Disabled Global Signalling Settings: --- Codecs: 0x8000e (gsm|ulaw|alaw|h263) Codec Order:none T1 minimum: 100 No premature media: No Relax DTMF: No Compact SIP headers:No RTP Keepalive: 0 (Disabled) RTP Timeout:0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Default Settings: - Context:default Nat:RFC3581 DTMF: rfc2833 Qualify:0 Use ClientCode: No Progress inband:Never Language: (Defaults to English) MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk Parsing /etc/asterisk/extconfig.conf sip show peer * Name : 155 Secret :Set MD5Secret:Not set Context : extern Language : en AMA flags: Unknown Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : Glen's Sysadmin Test Line200111222 ACL : No Codec Order : (none) Auto-Framing: No -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way voice with Asterisk
Hello All; I have more clues that may assist in resolving this: If I use the same softphone and dial out with the same Asterisk server. The SIP/voice traffic is able to be heard in both directions. So, anyone have any ideas for me? Still a little clueless. Glen On 11/6/2010 13:00, Zuhair Raza wrote: Hi Try Nat=yes in general settings On 06-Nov-2010 9:57 PM, Silver Thorne zora...@gmail.com mailto:zora...@gmail.com wrote: Let me explain: When I dial into Asterisk ( I have a SIP trunk - which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing - the extension rings on the Asterisk server (you can see it in the CLI and hear it at the computer), and the softphone rings However, when you answer the SIP softphone , you can only hear the voice FROM the softphone out. Where would I start to troubleshoot this? I am a little clueless! Thanks for all of your help. Asterisk 1.4.31 built by root @ some_server.foo.net http://some_server.foo.net on a x86_64 running Linux on 2010-06-10 14:32:34 UTC Sip Settings: Global Settings: SIP Port: 5060 Bindaddress: 0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Promsic. redir: No SIP domain support: No Call to non-local dom.: Yes URI user is phone no: No Our auth realm asterisk Realm. auth: No Always auth rejects: No Call limit peers only: No Direct RTP setup: No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off IP ToS SIP: none IP ToS RTP audio: none IP ToS RTP video: none T38 fax pt UDPTL: No RFC2833 Compensation: No SIP realtime: Disabled Global Signalling Settings: --- Codecs: 0x8000e (gsm|ulaw|alaw|h263) Codec Order: none T1 minimum: 100 No premature media: No Relax DTMF: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Default Settings: - Context: default Nat: RFC3581 DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: (Defaults to English) MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk Parsing /etc/asterisk/extconfig.conf sip show peer * Name : 155 Secret :Set MD5Secret :Not set Context : extern Language : en AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup : Pickupgroup : Callerid : Glen's Sysadmin Test Line200111222 ACL : No Codec Order : (none) Auto-Framing: No -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple extensions - same context
Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn! I have several users configured (101, 102, 105, 155, 211, etc). They are all in different contexts. I don't want to mess up what is already there, I just want to make sure that they are all able to dial each other. I would assume that they would all have to be in the same contexts. So, on that note, where would I look about information on putting all of my users in a common context without messing up what I already have. Do you follow? I hope so! Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with asterisk
:35] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. So, when I call the 33173793697 number, the above entry is what I see in the log. Glen On 11/1/2010 17:32, Steve Edwards wrote: On Mon, 1 Nov 2010, Silver Thorne wrote: Anyone see this before: [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have 6839, digest has3169 You may have better luck with a more descriptive subject. Lots of users have an issue or two with Asterisk. Some details will also help. Like: ) Version of Asterisk. ) Name and version of the endpoints involved. ) Relevant sections of sip.conf as well as the console output from 'sip show settings,' 'sip show userusername,' and 'sip show peer peername.' (I'm a 1.2 Luddite.) ) Console output of 'sip debug ipaddress' illustrating the 'issue.' Don't forget to 'sanitize' any IP addresses, usernames, and passwords that you consider valuable. (Actually, it would be better to redo your configuration with 'throw-away' credentials (like username1 and password1) for the duration of your issue -- less chance of exposing something or mistyping an important detail.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with asterisk
Hey; I never thought of that. It is causing an issue for me. One SIP UA works fine - ring, forward, etc. While the other does not. I am a little clueless here - where would I start with this? Thanks Glen On 11/1/2010 19:15, Philipp von Klitzing wrote: Hi! [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have 6839, digest has3169 You most likely have two SIP UAs that use the same IP, of which the 6839 account is listed last in sip.conf while 3169 is trying to auth (unsuccessfully). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX or SIP - connecting two Asterisk servers together
Hello Folks; Again, excuse my cluelessness. I have an Asterisk server in the US - and I want to connect it to one in Europe. Here is my scenario: 1. call a phone number, my Asterisk box in the US answers 2. perhaps a 'please wait' voice message 3. it dials an extension on the other Asterisk box in Europe. I am not looking for someone to do this for me, I am just not really sure how to get started. Perhaps some suggested reading, examples, etc? Any help at all would be appreciated. Thanks much. Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with asterisk
Hey; Anyone see this before: [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have 6839, digest has 3169 G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
