[asterisk-users] Callwithus.com is discontinuing IAX service
Callwithus.com is discontinuing iax service. Can anybody recommend IAX provider - I need somebody with good rates to Philippines. -- Joseph ___ -- 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] about monitored calls storing
Hello all, how can I possibly make the monitoring for all calls through the asterisk, and for those file to be stored with the name of the initiator, in additional to know to whom this call is going, could this functionality be implemented via configurations! in other words, could I configure the asterisk so that the administrator to be able to hear calls coming from who going to whom, as a having a record for each call, I am using trixbox v2.6.2.1 should that functionality be implemented by an external application , such as one written using asterisk-java !!! any help is appreciated? thanks in advance, ___ -- 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] underlying sound during sip calls
Hi we have set up two asterisk machines where we do all the managing of SIP calls. Now, sometimes we call and we get an underlying sound that is a locution from a customer. What could make this to happen ? Is very strange to us, but maybe we are missing something... some configuration, or anything else . . . . Thank you so much. ___ -- 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] snom mass deploy help
Sorry for dropping in so late, but maybe our solution do configure snom phones can help you We have written a small script that scans the network for snom phones. This is done by doing broadcast pings and using the arp-scan command and reading the arp cache. Then we filter the results on the mac addresses that start with 00:04:13. Of course there are other solutions to get the list of IP addresses of your snom phones. Once we have a list of IP addresses we push our configuration url to the snom phones by calling a url on the phones. http://$IP/dummy.htm?settings=savesetting_server=$massdeploymenturl DHCP was not an option for us because we did not want to change or interfere with the customer existing network structure. Best regards, Loïc Didelot. On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote: Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex ___ -- 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 -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.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
Re: [asterisk-users] Sangoma A200
Alex Samad wrote: Voltage isn't the issue - the difference is in the impedance. Australia I get this in my dmesg when I load up the rdm410 modules [1083334.103487] Freed a Wildcard [1083336.171371] ALAW override parameter detected. Device will be operating in ALAW [1083338.040522] Boosting ringer on slot 1 (89V peak) [1083338.040542] Port 1: Installed -- AUTO FXS/DPO [1083340.340472] Boosting ringer on slot 2 (89V peak) [1083340.340492] Port 2: Installed -- AUTO FXS/DPO ALAW is good - that's the default in use for Australia. FXS ports aren't relevant since they're for extensions, not PSTN lines. uses complex impedance (220+820Ohm resistors with a 120nF capacitor) whereas the US uses a straight resistor. Did yo buy from the us or local ? Local I believe, though I can't tell you through whom. (I didn't handle the purchase of the card - merely the installation) ___ -- 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] unwanted locution
Hi, we are experiencing a problem that is very strange, only on SOME calls, a locution jumps in to the RTP stream and both persons between the phones can hear it. It is looped and it does not stop till hang up. Do you have any clue about what could be happening ? Thank you ! ___ -- 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] about monitored calls storing
Trixbox I think uses FreePBX FreePbx has an option for each extension to set it to record all calls. It will record the extension in the file name and you can view it through the recordings app if you want a web view. There are all stored in a common dir /var/spool/asterisk/monitor - you can probably mod the code for the recording if you want more info in the filename Cheers Duncan peace keeper wrote: Hello all, how can I possibly make the monitoring for all calls through the asterisk, and for those file to be stored with the name of the initiator, in additional to know to whom this call is going, could this functionality be implemented via configurations! in other words, could I configure the asterisk so that the administrator to be able to hear calls coming from who going to whom, as a having a record for each call, I am using trixbox v2.6.2.1 should that functionality be implemented by an external application , such as one written using asterisk-java !!! any help is appreciated? thanks in advance, ___ -- 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] unwanted locution
Please post ONCE to the list. Please define the 'locution'. What words can you hear? (BTW locution is a rather uncommonly used word in english). Steve On 29 Jun 2009, at 10:07, Xavier Cardil wrote: Hi, we are experiencing a problem that is very strange, only on SOME calls, a locution jumps in to the RTP stream and both persons between the phones can hear it. It is looped and it does not stop till hang up. Do you have any clue about what could be happening ? Thank you ! ___ -- 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] how to sniff RTP and SIP traffic only
Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- 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] unwanted locution
I meant that we hear an audio stream typical from an IVR application, that says press 1 if you want to . . blabla press two if you want to . . . We have checked the configuration and the code of our IVR application but we can't see why this is playing, and only in some calls, not in all calls. We don't know about the procedence of that audio stream . . . Sorry for the double post but I thought it wasn't sent. Thank you. On Mon, Jun 29, 2009 at 11:43 AM, Steve Howes st...@geekinter.net wrote: Please post ONCE to the list. Please define the 'locution'. What words can you hear? (BTW locution is a rather uncommonly used word in english). Steve On 29 Jun 2009, at 10:07, Xavier Cardil wrote: Hi, we are experiencing a problem that is very strange, only on SOME calls, a locution jumps in to the RTP stream and both persons between the phones can hear it. It is looped and it does not stop till hang up. Do you have any clue about what could be happening ? Thank you ! ___ -- 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 ___ -- 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] how to sniff RTP and SIP traffic only
For Linux use tcpdump on the host you are after tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0 where 5060 is your SIP port and 1-16000 are your rtp ranges -s0 means snap length of 0 so capture all the packet rather than cutting off at a point And refine it by adding the host you are targetting and -w to write to a file. Then you can import the file in wireshark and use the voip utlities to listen to it fairly easily or use tcpdump -r to read it back and clean it out a bit more Cheers Duncan Xavier Cardil wrote: Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- 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] how to sniff RTP and SIP traffic only
Thank you so much ! On Mon, Jun 29, 2009 at 12:21 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: For Linux use tcpdump on the host you are after tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0 where 5060 is your SIP port and 1-16000 are your rtp ranges -s0 means snap length of 0 so capture all the packet rather than cutting off at a point And refine it by adding the host you are targetting and -w to write to a file. Then you can import the file in wireshark and use the voip utlities to listen to it fairly easily or use tcpdump -r to read it back and clean it out a bit more Cheers Duncan Xavier Cardil wrote: Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- 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 ___ -- 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] unwanted locution
On 29 Jun 2009, at 11:00, Xavier Cardil wrote: I meant that we hear an audio stream typical from an IVR application, that says press 1 if you want to . . blabla press two if you want to . . . We have checked the configuration and the code of our IVR application but we can't see why this is playing, and only in some calls, not in all calls. We don't know about the procedence of that audio stream . . . Sorry for the double post but I thought it wasn't sent. Does it sound like *your* IVR? If not, are you using analogue lines anywhere? S ___ -- 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] unwanted locution
No, it doesn't sound like any of our IVR applications. We have 2 two Cisco AS5400 PSTN to SIP gateways, so yes we do outbound / inbound calls. Thank you. On Mon, Jun 29, 2009 at 12:52 PM, Steve Howes st...@geekinter.net wrote: On 29 Jun 2009, at 11:00, Xavier Cardil wrote: I meant that we hear an audio stream typical from an IVR application, that says press 1 if you want to . . blabla press two if you want to . . . We have checked the configuration and the code of our IVR application but we can't see why this is playing, and only in some calls, not in all calls. We don't know about the procedence of that audio stream . . . Sorry for the double post but I thought it wasn't sent. Does it sound like *your* IVR? If not, are you using analogue lines anywhere? S ___ -- 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] unwanted locution
My guess would be the fault is on the analogue side not the SIP. Steve On 29 Jun 2009, at 12:05, Xavier Cardil wrote: No, it doesn't sound like any of our IVR applications. We have 2 two Cisco AS5400 PSTN to SIP gateways, so yes we do outbound / inbound calls. Thank you. On Mon, Jun 29, 2009 at 12:52 PM, Steve Howes st...@geekinter.net wrote: On 29 Jun 2009, at 11:00, Xavier Cardil wrote: I meant that we hear an audio stream typical from an IVR application, that says press 1 if you want to . . blabla press two if you want to . . . We have checked the configuration and the code of our IVR application but we can't see why this is playing, and only in some calls, not in all calls. We don't know about the procedence of that audio stream . . . Sorry for the double post but I thought it wasn't sent. Does it sound like *your* IVR? If not, are you using analogue lines anywhere? S ___ -- 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 ___ -- 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] Skype for Asterisk. Any return of experience ?
I received a phone call asking for specs, how I'd use, etc, etc and they said they'd be turning my beta account up in another 6 weeks. That was 3 weeks ago. PB On Mon, Jun 29, 2009 at 1:31 AM, randulo spamsucks2...@gmail.com wrote: Though they have written me back twice to say coming soon I am still waiting for the software... So you'd rather have it even when it hasn't been finished? Umm, no, but then when a company says looking for beta testers - please sign up now! and then four months later has nothing to let me beta test, I am a bit put off. The beta was limited. Digium wants to open it but says Skype themselves are delaying the operation. I have compelling reasons to believe this, even though I can't put them out in public. I was surprised too at the apparent slowness, but I think it will happen in good time. ___ -- 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] asterisk 1.4.21.2 a caller waited in queue, after connect to agent hears silence
Hi all!
My problem is that calls being placed in the queue, and are waiting
while the agents are busy, when an
agents is then free they gets connected to the agent but there is
silence (no voice).
If a caller has not to wait in the queue, there is no problem.
My agents have an iax2 client, and imcoming calls are over SIP.
queue.conf:
persistentmembers=yes
autofill=yes
ringinuse=no
eventwhencalled=yes
eventmemberstatus=yes
[servicenr1_w1]
musiconhold=default
strategy=leastrecent
ringinuse=no ; verhindert dass Telefone angerufen werden die sowieso
telefonieren
wrapuptime=5
timeout=20
retry=5
weight=1
autopause=yes
setinterfacevar=yes ;variable MEMBERINTERFACE wird gesetzt (zB. AGENT/21)
monitor-format = wav49
monitor-type = MixMonitor
joinempty=strict
leavewhenempty=strict
maxlen=20
member = Agent/21
member = Agent/22
member = Agent/23
The agents login with AgentCallbacklogin wiht
AgentCallBackLogin(21||*...@agents);
agents-context in extension.ael looks like this:
context agents
{
// Anrufe von queue an agenten
*21 = queuecallagent(${EXTEN:1});
*22 = queuecallagent(${EXTEN:1});
*23 = queuecallagent(${EXTEN:1});
}
macro queuecallagent(agentnr)
{
Dial(IAX2/${agentnr}||xtTg);
switch(${DIALSTATUS})
{
case BUSY:
break;
case ANSWER:
break;
default:
Hangup();
break;
}
}
agents.conf:
[general]
persistentagents=yes
[agents]
maxlogintries=3
musiconhold = default
updatecdr=yes
agent = 21,1234,Klaudia
agent = 22,1234,Daniele
agent = 23,1234,Daniela
What about the wrapuptime in agents.conf - do i have to set this to
the same as in queues.conf?
i also get this message:
WARNING[23115]: app_queue.c:3014 try_calling: The device state of
this queue member, Agent/22, is still 'Not in Use' when it probably
should not be! Please check UPGRADE.txt for correct configuration
settings.
the Agent/22 is not in use, there are no open channels and queue
show is also reporting not in use. So why i m getting this Warning?
i really don't know whats the reason of the silence.
yours
christian gansberger
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[asterisk-users] Transfer dropping calls
When doing transfers the call drops as follows: 1. I receive a call (internal or not) 2. I dial *2, wait for transfer sound plus dialtone 3. I dial for destinantion person, who pickups the phone 4. We talk to each other 5. I hangup my phone and the call drops if I dial * when talking with destination person a got the original call back The same occurs with blind transfers We are using Asterisk 1.4.25.1 and X-lite softphones. Thanks people Valter ___ -- 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] Calling non-extension numbers issue
Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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] Dail in modem
Sorry i replied late bcz i have to do some other work I have a new required functionality. that is Develop a Client server application that will communicate using a normal modem with out connecting to internet.(Client with a PC and modem will dail the number of server it will be a PSTN number (Not an ISP like thing) and the server with modem will recieve the call and receive some data and return results). Direct communication like hyper terminal. no connection to internet. i have tried TAPI(C#) and JTAPI (java) but dont get sucess. I am thinking Asterisk can handle that using TDM 400P card regards Shakeel Abbas On Sat, Jun 20, 2009 at 7:19 PM, Geraint Leegera...@gmail.com wrote: If i understand correctly you need users to be able to dial in using a modem to your servers then you are going to share your internet connection with those who dial your server. So, no, it has nothing to do with asterisk... you want to be looking at wvdial for the clients (assuming they are linux) and whatever the equivalent server would be (don't know as i've never done it). Good luck 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Geraint lee I also dont know .what kind of requirements are these :P i am just looking if it can happen On Fri, Jun 19, 2009 at 9:33 PM, Geraint Leegera...@gmail.com wrote: is it just me or am i right in thinking this has nothing to do with asterisk? 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- 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 ___ -- 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 ___ -- 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] Calling non-extension numbers issue
You have tried putting # after the number (for example 5551212#)? You could have a dialplan problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 8:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling non-extension numbers issue Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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] Calling non-extension numbers issue
I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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] Force Authentication
Hi all, i would like to ask please about how to force asterisk to ask for authentication when receiving an INVITE packet from any device? Regards ___ -- 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] Calling non-extension numbers issue
That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com mailto:ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 ___ -- 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] ISP -Asterisk - ATA -DIALUP
Hellow, * I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? * ISP - Asterisk - ATA - DIALUP -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased ___ -- 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] Calling non-extension numbers issue
IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 ___ -- 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] Calling non-extension numbers issue
This is the configured pattern for local calls - _NXXNXX When I dial a local number from the device, I dial a number like 7706743900 and select internet call. Does what I dial not match the pattern? Danny Nicholas wrote: IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kayton Sapale *Sent:* Monday, June 29, 2009 9:35 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com mailto:ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 ___ -- 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] Calling non-extension numbers issue
Your pattern appears to be set up in anticipation of a leading digit. What happens if you dial either 17706743900 or 97706743900? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue This is the configured pattern for local calls - _NXXNXX When I dial a local number from the device, I dial a number like 7706743900 and select internet call. Does what I dial not match the pattern? Danny Nicholas wrote: IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 _ ___ -- 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] Force Authentication
It does this by default unless you have allowguest set to yes, and/or any insecure parameter options on any individual peers. -- Sent from mobile device On Jun 29, 2009, at 10:33 AM, michel freiha mich...@gmail.com wrote: Hi all, i would like to ask please about how to force asterisk to ask for authentication when receiving an INVITE packet from any device? Regards ___ -- 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] Calling non-extension numbers issue
I tried the following: 17706743900 97706743900 On both, I receive the same timeout message. I also tried with different numbers as well, just to give a try, same results. Just for giggles, I tried: _7706743900 When trying this, I get a Address not in use message, which I think means that the number is interpreted as an extension, which is not configured. Danny Nicholas wrote: Your pattern appears to be set up in anticipation of a leading digit. What happens if you dial either 17706743900 or 97706743900? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kayton Sapale *Sent:* Monday, June 29, 2009 9:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Calling non-extension numbers issue This is the configured pattern for local calls - _NXXNXX When I dial a local number from the device, I dial a number like 7706743900 and select internet call. Does what I dial not match the pattern? Danny Nicholas wrote: IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kayton Sapale *Sent:* Monday, June 29, 2009 9:35 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com mailto:ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 ___ -- 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 ___ -- 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] ISP -Asterisk - ATA -DIALUP
Without getting into a lot of detail, this will not work. Period. You just can't do reliable modem passthrough with VoIP in most cases, some clever proprietary hacks notwithstanding. To the extent it is possible, nobody is going to send you the procedure.. This list is for specific answers to specific questions. -- Sent from mobile device On Jun 29, 2009, at 10:47 AM, Vidura Senadeera vidura...@gmail.com wrote: Hellow, I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? ISP - Asterisk - ATA - DIALUP -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased ___ -- 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] Calling non-extension numbers issue
One to few X's for that number? Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, June 29, 2009 10:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calling non-extension numbers issue Your pattern appears to be set up in anticipation of a leading digit. What happens if you dial either 17706743900 or 97706743900? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue This is the configured pattern for local calls - _NXXNXX When I dial a local number from the device, I dial a number like 7706743900 and select internet call. Does what I dial not match the pattern? Danny Nicholas wrote: IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 _ ___ -- 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] Calling non-extension numbers issue
After reading the Asterisk PDF again, your pattern is fine, it's probably just not in the right place in your dialplan. Check your sip.conf and users.conf to insure that the device hits the correct context to dial out. You might want to post them here for a quick checkup. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue I tried the following: 17706743900 97706743900 On both, I receive the same timeout message. I also tried with different numbers as well, just to give a try, same results. Just for giggles, I tried: _7706743900 When trying this, I get a Address not in use message, which I think means that the number is interpreted as an extension, which is not configured. Danny Nicholas wrote: Your pattern appears to be set up in anticipation of a leading digit. What happens if you dial either 17706743900 or 97706743900? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue This is the configured pattern for local calls - _NXXNXX When I dial a local number from the device, I dial a number like 7706743900 and select internet call. Does what I dial not match the pattern? Danny Nicholas wrote: IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- 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 _ ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
Re: [asterisk-users] registration failed, not a local domain
jonas kellens wrote: asterisk*CLI sip show domains Our local SIP domains: Context Set by jocan.local (default) [Configured] 192.168.1. (default) [Configured] [Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889 handle_request_register: Registration from 'sip:grandstr...@192.168.1.248' failed for '192.168.1.13' - Not a local domain SIP.conf : domain=jocan.local ; Add IP address as local domain domain=192.168.1. ; You can have several domain settings I've tried : domain=192.168.1.0 domain=192.168.1 My SIP-phones are in the subnet 192.168.1.0/255.255.255.0, my Asterisk has 192.168.1.248 You want to configure the IP adress 192.168.1.248 (the IP of the Asterisk box) in sip.conf: domain=192.168.1.248 Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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] T38 Fax Gateway for Asterisk 1.6
James Lamanna wrote: I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. You might be looking for this issue in the bug tracker: https://issues.asterisk.org/view.php?id=13405 Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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] CRMy type app?
Looking for a (windows) app. that will listen to the manager interface then pop-up a web browser pointing to a page on an incoming phone call.. Not looking for outlook integration, or outbound dialling, just to recognise an incoming call and poke a URL at a website in a browser and I've absolutely no idea how to do it in the MS windows world... Any clues appreciated.. (More pointing to an existing app. rather than how to write it myself!) Thanks, Gordon ___ -- 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] Callwithus.com is discontinuing IAX service
On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote: Callwithus.com is discontinuing iax service. Can anybody recommend IAX provider - I need somebody with good rates to Philippines. -- Joseph ___ -- 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 Voipstreet.com seems to have very good service. I have been testing for several weeks now, without issue. -- A.G. (Tony) Nichols I.S. Manager ___ -- 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] Callwithus.com is discontinuing IAX service
On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote: Callwithus.com is discontinuing iax service. Can anybody recommend IAX provider - I need somebody with good rates to Philippines. -- Joseph Did you ask why they are dropping IAX2? That is a fairly big decision. I assume it was eating more resources than it was worth. Sort of like IAX.cc (Vitelity) recommending IAX2. I have fixed many ITPS setup by simple switching to SIP. Just some questions to ask, and answers to ponder. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] 2 problems I can't solve without any help
jonas kellens wrote: Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : I have the same (or similar) issue with one of my ITSPs. In my case, the problem seemed to be because they do some load balancing: after turning on sip debug and making numerous incoming calls, I noticed that the calls come from a number of different IP addresses. The call was sent to the proper context (from-itsp,s,1) when the incoming IP address matched the host=ip.ad.re.ss in sip.conf - else it went to the default (default,s,1) I don't think host= allows multiple hosts to be defined, so I tried creating a new sip.conf entry for every IP address observed. It worked, but is nasty looking and doesn't scale. Instead I ended up doing: sip.conf [general] context=default register = mynumber:mypassw...@ip.ad.re.ss/itspname [itsp] ; stuff recommended in ITSP's documentation for asterisk configuration -=-=- extensions.conf [default] exten = itspname,1,NoOp(incoming call from itsp) exten = itspname,n,Goto(from-itsp,s,1) [from-itsp] exten = s,1,Dial(SIP/myphone) ; Need this for the times when the itsp calls from the same IP address as defined in sip.conf host= line exten = itspname,1,Goto(from-itsp,s,1) -=-=- YMMV depending on how your ITSP handles the /itspexten in your register = statement.You might need to use your account's number. Problem 2 Setup : Grandstream -- Asterisk -- Endian_Firewall -- SIPprovider Problem : Called party can not here me (I'm on the Grandstream) while I can here the other side clearly (GSM/cell phone number). First - does audio between Asterisk and the Grandstream work symmetrically? You can test it with any of: - Phone another SIP client on the LAN (softphone like Zoiper will work if you don't have another hardphones around) - use echotest [exten = 555,1,Echo()] - record/playback an audio file (voicemail, or Record()/Playback() ) I am not familiar with Endian, find out what it does for logging and crank the verbosity up so you can see what packets are being accepted/dropped/never arrive. firewall : -A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT With -A (as opposed to -I) any already existing rules that would drop this traffic will take precedence. Since you are already behind the Endian_firewall - flush your firewall rules on the asterisk box and set default policies to accept. (lock it down again later after you've got the audio working) Configuration Endian : portforwarding : 5060 and 11000:11500 to Asterisk_internal_ip This should be okay, watch the logging for any traffic coming from your ITSP's IP address(es) that is being dropped. outgoing traffic : coming from Asterisk_internal_ip : ports 5060 and 11000:11500 to RED ZONE (internet) are open ! 11000-11500 is your own local rtp port range, not the ITSP's port range. You can control what ports are used locally, but you have no control over what ports are available/used remotely. If your ITSP is using, for example, 2-5 then your audio will never be heard on the far end because of this firewall rule. Check your ITSP's support/documentation to see what port range they use, or allow ALL outgoing traffic from Asterisk_internal_ip. ___ -- 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] OT: Mobile voip - WCell
Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia next month. Only problem is $10 minimum to get the account started. Any thoughts or is it crap and the att data service doesn't work well enough to use this. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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] OT: Mobile voip - WCell
URL reeks of affiliate link. http://www.wcell.com/index.php gets you to the same place your /ab/ZnJpZW5kLzEwNjI http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= affiliate link does. Not sure how you navigate to get the URL you posted. Affiliate = commercial Thanks, Steve Totaro On Mon, Jun 29, 2009 at 2:13 PM, Dean Collins d...@cognation.net wrote: Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia next month. Only problem is $10 minimum to get the account started. Any thoughts or is it crap and the att data service doesn’t work well enough to use this. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] OT: Mobile voip - WCell
http://www.wcell.com/incentives.php MLM, uuggg I have some Amway stuff you can buy too! On Mon, Jun 29, 2009 at 2:26 PM, Steve Totaro stot...@totarotechnologies.com wrote: URL reeks of affiliate link. http://www.wcell.com/index.php gets you to the same place your /ab/ZnJpZW5kLzEwNjI http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= affiliate link does. Not sure how you navigate to get the URL you posted. Affiliate = commercial Thanks, Steve Totaro On Mon, Jun 29, 2009 at 2:13 PM, Dean Collins d...@cognation.net wrote: Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia next month. Only problem is $10 minimum to get the account started. Any thoughts or is it crap and the att data service doesn’t work well enough to use this. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] CRMy type app?
On Mon, 29 Jun 2009, Gordon Henderson wrote: Looking for a (windows) app. that will listen to the manager interface then pop-up a web browser pointing to a page on an incoming phone call.. Not looking for outlook integration, or outbound dialling, just to recognise an incoming call and poke a URL at a website in a browser and I've absolutely no idea how to do it in the MS windows world... Any clues appreciated.. (More pointing to an existing app. rather than how to write it myself!) Following up my own post - found ADAT at http://www.tttelecom.nl/index.php?option=com_contentview=articleid=23Itemid=7lang=en Seems to fit the bill... Anyone else using it? Gordon ___ -- 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] snom mass deploy help
2009/6/29 Loic Didelot ldide...@mixvoip.com Sorry for dropping in so late, but maybe our solution do configure snom phones can help you We have written a small script that scans the network for snom phones. This is done by doing broadcast pings and using the arp-scan command and reading the arp cache. Then we filter the results on the mac addresses that start with 00:04:13. Of course there are other solutions to get the list of IP addresses of your snom phones. So, DHCP server has already replied to Snom's address request, isn't it ? Once we have a list of IP addresses we push our configuration url to the snom phones by calling a url on the phones. How do you call a url ? Is there a menu or key dedicated to that ? http://$IP/dummy.htm?settings=savesetting_server=$massdeploymenturl What happens to phones before this url is called (and after they get an address) ? Are they usable for something ? Myself, I was thinking of using a 2 steps configuration process with an IVR allowing a phone to go from Connected to Configured status. DHCP was not an option for us because we did not want to change or interfere with the customer existing network structure. Best regards, Loïc Didelot. On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote: Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex ___ -- 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 -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.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 ___ -- 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] OT: Mobile voip - WCell
Just found out from Mitchel Constantin via a Facebook reply that his company installed the Freeswitch backend backend for Wcell. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Monday, June 29, 2009 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Mobile voip - WCell http://www.wcell.com/incentives.php MLM, uuggg I have some Amway stuff you can buy too! On Mon, Jun 29, 2009 at 2:26 PM, Steve Totaro stot...@totarotechnologies.com wrote: URL reeks of affiliate link. http://www.wcell.com/index.php gets you to the same place your /ab/ZnJpZW5kLzEwNjI http://www.wcell.com/ab/ZnJpZW5kLzEwNjI = affiliate link does. Not sure how you navigate to get the URL you posted. Affiliate = commercial Thanks, Steve Totaro On Mon, Jun 29, 2009 at 2:13 PM, Dean Collins d...@cognation.net wrote: Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia next month. Only problem is $10 minimum to get the account started. Any thoughts or is it crap and the att data service doesn't work well enough to use this. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Asterisk ended with exit status 134... Asterisk exited on signal 6.
I was trying to enable CDR in a mysql-database when the following
occured :
asterisk*CLI cdr status
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: yes
CDR registered backend: cdr_manager
CDR registered backend: cdr-custom
asterisk*CLI exit
[Jun 29 21:56:52] Executing last minute cleanups
[r...@asterisk asterisk]# vi modules.conf
[r...@asterisk asterisk]# service asterisk restart
Shutting down asterisk: [ OK ]
Starting asterisk: [ OK ]
[r...@asterisk asterisk]# *** glibc detected *** /usr/sbin/asterisk:
corrupted double-linked list: 0x08eedb18 ***
=== Backtrace: =
/lib/libc.so.6[0x17fda9]
/lib/libc.so.6[0x18192d]
/lib/libc.so.6(__libc_calloc+0xef)[0x18348f]
/usr/lib/asterisk/modules/chan_sip.so[0x86d1b1]
/usr/lib/asterisk/modules/chan_sip.so[0x87fcaf]
/usr/lib/asterisk/modules/chan_sip.so[0x89416c]
/usr/sbin/asterisk(ast_sched_runq+0xd3)[0x80f3a23]
/usr/lib/asterisk/modules/chan_sip.so[0x89382e]
/usr/sbin/asterisk[0x81014fb]
/lib/libpthread.so.0[0xa3e51f]
/lib/libc.so.6(clone+0x5e)[0x1f404e]
=== Memory map:
0011-0027e000 r-xp fd:00 309034 /lib/libc-2.9.so
0027e000-0028 r--p 0016e000 fd:00 309034 /lib/libc-2.9.so
0028-00281000 rw-p 0017 fd:00 309034 /lib/libc-2.9.so
00281000-00284000 rw-p 00281000 00:00 0
00284000-0028a000 r-xp fd:00
122266 /usr/lib/asterisk/modules/res_adsi.so
0028a000-0028b000 rw-p 5000 fd:00
122266 /usr/lib/asterisk/modules/res_adsi.so
0028b000-00292000 r-xp fd:00
122275 /usr/lib/asterisk/modules/res_musiconhold.so
00292000-00293000 rw-p 7000 fd:00
122275 /usr/lib/asterisk/modules/res_musiconhold.so
00293000-0029d000 r-xp fd:00 309038 /lib/libcrypt-2.9.so
0029d000-0029e000 r--p 9000 fd:00 309038 /lib/libcrypt-2.9.so
0029e000-0029f000 rw-p a000 fd:00 309038 /lib/libcrypt-2.9.so
0029f000-002c6000 rw-p 0029f000 00:00 0
002c6000-002c8000 r-xp fd:00 309082 /lib/libcom_err.so.2.1
002c8000-002c9000 rw-p 1000 fd:00 309082 /lib/libcom_err.so.2.1
002c9000-002d2000 r-xp fd:00
50962 /usr/lib/libkrb5support.so.0.1
002d2000-002d3000 rw-p 8000 fd:00
50962 /usr/lib/libkrb5support.so.0.1
002d3000-002d6000 r-xp fd:00
122298 /usr/lib/asterisk/modules/app_authenticate.so
002d6000-002d7000 rw-p 2000 fd:00
122298 /usr/lib/asterisk/modules/app_authenticate.so
002d7000-002d9000 r-xp fd:00
122315 /usr/lib/asterisk/modules/app_exec.so
002d9000-002da000 rw-p 1000 fd:00
122315 /usr/lib/asterisk/modules/app_exec.so
002da000-002dc000 r-xp fd:00
122392 /usr/lib/asterisk/modules/func_audiohookinherit.so
002dc000-002dd000 rw-p 1000 fd:00
122392 /usr/lib/asterisk/modules/func_audiohookinherit.so
002dd000-002e1000 r-xp fd:00
122274 /usr/lib/asterisk/modules/res_monitor.so
002e1000-002e2000 rw-p 4000 fd:00
122274 /usr/lib/asterisk/modules/res_monitor.so
002e2000-00352000 r-xp fd:00
55924 /usr/lib/libodbc.so.1.0.0
00352000-00357000 rw-p 0006f000 fd:00
55924 /usr/lib/libodbc.so.1.0.0
00357000-0036d000 r-xp fd:00 309044 /lib/libnsl-2.9.so
0036d000-0036e000 r--p 00016000 fd:00 309044 /lib/libnsl-2.9.so
0036e000-0036f000 rw-p 00017000 fd:00 309044 /lib/libnsl-2.9.so
0036f000-00371000 rw-p 0036f000 00:00 0
00371000-0037c000 r-xp fd:00
309050 /lib/libnss_files-2.9.so
0037c000-0037d000 r--p a000 fd:00
309050 /lib/libnss_files-2.9.so
0037d000-0037e000 rw-p b000 fd:00
309050 /lib/libnss_files-2.9.so
0037e000-00382000 r-xp fd:00
122336 /usr/lib/asterisk/modules/app_playback.so
00382000-00383000 rw-p 3000 fd:00
122336 /usr/lib/asterisk/modules/app_playback.so
00383000-00384000 r-xp fd:00
122314 /usr/lib/asterisk/modules/app_echo.so
00384000-00385000 rw-p fd:00
122314 /usr/lib/asterisk/modules/app_echo.so
00385000-00389000 r-xp fd:00
122316 /usr/lib/asterisk/modules/app_externalivr.so
00389000-0038a000 rw-p 3000 fd:00
122316 /usr/lib/asterisk/modules/app_externalivr.so
0038a000-0038b000 r-xp fd:00
122409 /usr/lib/asterisk/modules/func_rand.so
0038b000-0038c000 rw-p 1000 fd:00
122409 /usr/lib/asterisk/modules/func_rand.so
0038c000-00391000 r-xp fd:00
122276 /usr/lib/asterisk/modules/res_odbc.so
00391000-00392000 rw-p 4000 fd:00
122276 /usr/lib/asterisk/modules/res_odbc.so
00392000-004b5000 r-xp fd:00 101032 /usr/sbin/safe_asterisk:
line 125: 2072 Aborted (core dumped) nice -n $PRIORITY
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} /dev/${TTY}
/dev/${TTY}
Asterisk ended with exit status 134
Asterisk exited on signal 6.
Automatically restarting Asterisk.
mpg123: no process killed
Any feedback ?
Thanks !
Jonas.
___
-- Bandwidth and
Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. Thanks. -- James ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Why can't you just do a daily/weekly cron to restart when convenient in off/slow hours for local time. Is your business constantly on-line 24/7? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Monday, June 29, 2009 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phones dropping registration,but asterisk thinks phones are still registered On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. Thanks. -- James ___ -- 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] snom mass deploy help
On Mon, Jun 29, 2009 at 10:22:54AM +0200, Loic Didelot wrote: Sorry for dropping in so late, but maybe our solution do configure snom phones can help you We have written a small script that scans the network for snom phones. This is done by doing broadcast pings and using the arp-scan command and reading the arp cache. Then we filter the results on the mac addresses that start with 00:04:13. Of course there are other solutions to get the list of IP addresses of your snom phones. Once we have a list of IP addresses we push our configuration url to the snom phones by calling a url on the phones. http://$IP/dummy.htm?settings=savesetting_server=$massdeploymenturl DHCP was not an option for us because we did not want to change or interfere with the customer existing network structure. I have found this to be the simplest way to do it https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder Best regards, Loïc Didelot. On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote: Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex ___ -- 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 -- What I'm suggesting to you is, if you can't name the foreign minister of Mexico, therefore, you know, you're not capable of what you do. But the truth of the matter is you are, whether you can or not. - George W. Bush 11/06/1999 as quoted in the Seattle Post-Intelligencer signature.asc Description: Digital signature ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
On Mon, Jun 29, 2009 at 4:23 PM, James Lamanna jlama...@gmail.com wrote: On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? Yes, I've experienced the same thing. Not sure right now what Asterisk version I'm using, prob the latest in the Ubuntu 8.04 repos. Just my 2c, fwiw. -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- 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] how to sniff RTP and SIP traffic only
Xavier Cardil wrote: Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? I'm not sure what you mean by for a faster debugging. As for sniffing the traffic, tcpdump works well. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] snom mass deploy help
Hello, I think you do not understand. The phone gets an IP from the DHCP server. Nothing special in this step. Then you save the configuration url to the snom by calling the url I showed you in my previous e-mail. This can be done using: wget, lynx, links, php, perl .. Once the phone has the configuraiton url, it then takes the settings from this url. So it is up to you to write a script that returns the correct XML. Best regards, Loïc Didelot. On Mon, 2009-06-29 at 21:40 +0200, Olivier wrote: 2009/6/29 Loic Didelot ldide...@mixvoip.com Sorry for dropping in so late, but maybe our solution do configure snom phones can help you We have written a small script that scans the network for snom phones. This is done by doing broadcast pings and using the arp-scan command and reading the arp cache. Then we filter the results on the mac addresses that start with 00:04:13. Of course there are other solutions to get the list of IP addresses of your snom phones. So, DHCP server has already replied to Snom's address request, isn't it ? Once we have a list of IP addresses we push our configuration url to the snom phones by calling a url on the phones. How do you call a url ? Is there a menu or key dedicated to that ? http://$IP/dummy.htm?settings=savesetting_server= $massdeploymenturl What happens to phones before this url is called (and after they get an address) ? Are they usable for something ? Myself, I was thinking of using a 2 steps configuration process with an IVR allowing a phone to go from Connected to Configured status. DHCP was not an option for us because we did not want to change or interfere with the customer existing network structure. Best regards, Loïc Didelot. On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote: Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex ___ -- 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 -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.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 ___ -- 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] Calling Number Verification Number? for BellSouth/ATT
BellSouth (now ATT) has a number you can dial and it will play back voice prompts with your calling number? It's used by their techs with a buttset in identifying analog 1FB lines... Eg Dial 704-210-3233, it answers Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two if your dialing from 704-559-2122 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Callwithus.com is discontinuing IAX service
On 06/29/09 13:54, Steve Totaro wrote: On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote: Did you ask why they are dropping IAX2? That is a fairly big decision. I assume it was eating more resources than it was worth. Sort of like IAX.cc (Vitelity) recommending IAX2. I have fixed many ITPS setup by simple switching to SIP. Just some questions to ask, and answers to ponder. Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) --copy of the email I received from them Dear Customers, We got some operational cost cuts because of discontinuing IAX protocol support, and pass the savings to you. Call rates to USA, Canada, UK, India mobile and some other destinations are reduced. Updated call rates are on our web site. Hint: the link to downloadable rate list is at the bottom of \A-Z rates\ web page. We\'d like to remind you that you have to use sip.callwithus.com server to make calls. IAX customers could continue to use iax.callwithus.com server before Sptember 1st deadline. Have a nice calling:-) ---end copy- Is SIP really so much better than IAX2 that providers are dropping it? The major issue for me with SIP is firewall traversal. -- Joseph ___ -- 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] Calling Number Verification Number? forBellSouth/ATT
To clarify the question is what is the number for ATT Calling Number Verification? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Monday, June 29, 2009 7:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calling Number Verification Number? forBellSouth/ATT BellSouth (now ATT) has a number you can dial and it will play back voice prompts with your calling number? It's used by their techs with a buttset in identifying analog 1FB lines... Eg Dial 704-210-3233, it answers Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two if your dialing from 704-559-2122 Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Why can't you just do a daily/weekly cron to restart when convenient in off/slow hours for local time. Is your business constantly on-line 24/7? I have tried that. Unfortunately restart when convenient doesn't always seem to actually restart asterisk, presumably because there are stuck calls or something. Very annoying as well. -- James On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. ___ -- 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] Queue Issue (1.4.21.1)
Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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] Callwithus.com is discontinuing IAX service
On Mon, Jun 29, 2009 at 8:13 PM, Josephsyscon...@gmail.com wrote: Is SIP really so much better than IAX2 that providers are dropping it? The major issue for me with SIP is firewall traversal. I wouldn't say 'better' so much as 'understood'. When I've worked with SIP, including integrating components from multiple manufacturers or programmers, things with SIP have worked or at least my troubleshooting led to measurable progress until I had things working. I had much difficulty locating suggestions for what I might be doing wrong when my IAX connections failed, or seemed to spontaneously reconnect, etc. I gave up, and went with a SIP-based approach because I need 'something that worked' more than I needed to know the ins and outs of IAX. Maybe somebody who knows the common errors, what causes them, and how to fix them could write up an IAX 101 or something. I have a hunch that provider has noticed the same thing, such that they end up providing more in-person support than it was possible for the margin they were making on that business. ___ -- 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] Calling Number Verification Number? forBellSouth/ATT
provided by mci 800.444. Thanks, Steve Totaro On Mon, Jun 29, 2009 at 8:09 PM, Jason Aarons (US) jason.aar...@us.didata.com wrote: To clarify the question is what is the number for ATT Calling Number Verification? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jason Aarons (US) *Sent:* Monday, June 29, 2009 7:24 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Calling Number Verification Number? forBellSouth/ATT BellSouth (now ATT) has a number you can dial and it will play back voice prompts with your calling number? It’s used by their techs with a buttset in identifying analog 1FB lines… Eg Dial 704-210-3233, it answers “Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two” if your dialing from 704-559-2122 -- *Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. * -- * Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. * ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Queue Issue (1.4.21.1)
The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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] Restricting domains with SIP Trunking
Hello, all. We have successfully connected our new Asterisk 1.6.1.1 PBX to Vitelity's network and have been very happy with them thus far. However, we'd like to use domains in our sip.conf to facilitate routing in our multi-tenant environment. We also like to set allowexternaldomains=no for security. However, this breaks our inbound PSTN calling from Vitelity. Is it possible to use allowexternaldomains=no with VoIP carriers? We tried setting fromdomain=a local domain. We tried defining a Vitelity domain based upon both vitelity.net and IP address (although I'm really hesitant to do that in case it changes). None of it worked. I have explicitly defined my localhost by name and IP address although I don't think that matters for this problem. How are others doing this? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Queue Issue (1.4.21.1)
It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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] Resetting CDRs on inbound calls
Hello, all. I see that I can use the C flag in Dial() to reset the CDRs and plan to use this for outbound calling since our carrier does not bill us until the call is answered. We'd like to do this for inbound calls from the PSTN since we pay for inbound and outbound minutes. However, it doesn't seem to make sense to do this with Dial() because we have an automated attendant which answers the call with Answer() and can add many seconds to the call length. How does one reset the CDRs in this case? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Queue Issue (1.4.21.1)
The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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] cisco phone 7911
Hellow, I have cisco 7911 and 7906 worked with asterisk server. But i can not set the time and date for these phones. can any one tell me how can i set the time and date for these phone. Thanks mahboob ___ -- 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] ISP -Asterisk - ATA -DIALUP
Not true... You can provided you disable data compression (ATK0) on your modem. Reason? Because a codec is already compressed. Adding compression at the modem level to an already compressed bitstream == lost bits. I call all over the world all the time using asterisk/sip/ulaw with decent bit rates. Alex Balashov wrote: Without getting into a lot of detail, this will not work. Period. You just can't do reliable modem passthrough with VoIP in most cases, some clever proprietary hacks notwithstanding. To the extent it is possible, nobody is going to send you the procedure.. This list is for specific answers to specific questions. -- Sent from mobile device On Jun 29, 2009, at 10:47 AM, Vidura Senadeera vidura...@gmail.com mailto:vidura...@gmail.com wrote: Hellow, / I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? / ISP - Asterisk - ATA - DIALUP -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased ___ -- 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 ___ -- 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] Queue Issue (1.4.21.1)
I think the handling of this may have improved in later versions of Asterisk - is an upgrade an option? (I tested this with a newer version of Asterisk recently, and it behaved how you were hoping it would behave) PaulH Kev Szaszvari wrote: The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- 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 This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to
