Re: [Asterisk-Users] JIAX status
On 12 Jun 2006, at 02:02, Rubens Zupelli Filho wrote: HI, Anyone knows the current status of JIAXclient? I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. I tried to e-mail the author but seems that his account is no longer valid. I in need of a java IAX client that could be loaded as an applet. I know that is a lot of viable SIP alternatives, but due to NAT/Firewall restrictions use of IAX would be easier. I'm due to give a talk about our pure Java IAX applet at Astricon London. Alternatively, if you contact me off list, I can give you the commercial details and point you at a demo URL. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rxfax with Sirrix quad BRI
Hi All Has anyone had experience with rxfax on asterisk 1.2.x with a sirrix quad BRI card? Does it work with the Sirrix cards? Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] key pads
Hi I am using asterisk at home 2.6 And I am using addpac as a voip gateway ,since I press # it will allow me to transfer the call . Any one knows what I should press to make a call conference Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax
Any one know a software can receive fax from asterisk . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enable/disable user
Hi, I want to know how can I enable/disable a user which is already has an account in the system. Is there any flat that I can use to disable an user? Please advise. Thanks ,unplug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] David Choo/eServices/eSpore is overseas
On Mon, June 12, 2006 4:37, David Choo said: I will be out of the office starting 12/06/2006 and will not return until 17/06/2006. Dear Sir / Mdm, I'm currently travelling. During this period of time, I have minimal access to internet and email. As such, please be aware that I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. SNIP Tongue mode='in cheek' That is good to know! We will start monitoring your residence until we find an opportune moment to enter. We will then lend a hand in (re)moving the most precious of your things to a new address... /Tongue (Sorry, couldn't help myself!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - SOLVED - Trouble getting SMS working
Hi, I was running asterisk 1.0.7. Today I upgraded to 1.2.9.1. The SMS still wasn't working, BUT it was very different. this is the output from the console -- Executing Goto(SIP/phone1-487d, smsmorx|s|1) in new stack -- Goto (smsmorx,s,1) -- Executing Answer(SIP/phone1-487d, ) in new stack -- Executing Wait(SIP/phone1-487d, 1) in new stack -- Executing SMS(SIP/phone1-487d, 101|sa) in new stack -- SMS TX 93 00 6D -- SMS RX 92 01 02 6B -- SMS TX 93 00 6D -- SMS RX 92 01 02 6B -- SMS TX 93 00 6D -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E -- SMS TX 92 01 FF 6E == Spawn extension (smsmorx, s, 3) exited non-zero on 'SIP/phone1-487d' Next I tried Michael's suggestion and compiled the app_sms.so from the 1.2.7.1 source tree. IT WORKED :D -- Executing Goto(SIP/phone1-5988, smsmorx|s|1) in new stack -- Goto (smsmorx,s,1) -- Executing Answer(SIP/phone1-5988, ) in new stack -- Executing Wait(SIP/phone1-5988, 1) in new stack -- Executing SMS(SIP/phone1-5988, 101|sa) in new stack -- SMS TX 93 00 6D -- SMS RX 91 0F 01 19 03 81 01 F2 00 F1 06 D4 E2 94 0A 8A 01 F9 -- SMS TX 95 02 00 00 69 -- SMS RX 94 00 6C -- Executing System(SIP/phone1-5988, /tmp/smstest s 101 Office) in new stack -- Executing Hangup(SIP/phone1-5988, ) in new stack == Spawn extension (smsmorx, s, 5) exited non-zero on 'SIP/phone1-5988' The message can be found in the /var/spool/asterisk/sms/morx/ directory. I now have another problem with sending messages back to the phone, when I run: smsq -o0198339100 -q101 --mttx-channel sip/phone1 --ud test I get WARNING[15516]:pbx_spool.c:346 scan_service: Unable to open /var/spool/asterisk/outgoing/smsq.mttx.101.1: Permission denied, deleting WARNING[15516]:pbx_spool.c:388 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/smsq.mttx.101.1' I will try to debug this during the week. Thanks to you both for your help. Mick. Hey Mick, which version of asterisk are you using ? I've experienced problems after 1.2.8 with app_sms.so. It seams that the application is sending out an sms to the center. But this message was never recieved by my cellphone. To solve the problem i compiled app_sms.so from 1.2.7.1 source tree and moved it into the modules directory. Someone should contact a developer for that - since I experienced the problem I did not have time to track the problem down to it's origin. Hope this helps! Greetings, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fixed ring strategy
Hi can i set in a queue or group a ring strategy that rings for every call only to an extension? for example: in the queue A i have SIP/200 SIP/201 SIP/202 SIP/203 every call is forwarded to SIP/200 and other members can pickup the call with *8 is it possible? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TTS engine query
Doug Crompton wrote: Not being very happy with festival I would like ro get a better TTS engine. I looked at the listings at: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international but I would like to get user input on suggested packages for Linux. Best performance vs. cost I didn't see Cepstral in that list. www.cepstral.com This is commercial software, I just registered Diane. $29.95 USD. I also see that they charge for a currency license. But, if it works well, we'll probably get a few licenses. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TTS engine query
Doug Lytle wrote: Doug Crompton wrote: This is commercial software, I just registered Diane. $29.95 USD. I also see that they charge for a currency license. But, if it works well, we'll probably get a few licenses. That should have read *concurrency*. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] - SOLVED - Trouble getting SMS working
I now have another problem with sending messages back to the phone, when I run: smsq -o0198339100 -q101 --mttx-channel sip/phone1 --ud test You need to run smsq as the Asterisk user, or else the file is created with permissions that Asterisk can't read and/or move. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended transfer and queue
It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active.Does anyone know any workarounds for this problem? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - SOLVED - Trouble getting SMS working
All working great now! Cheers. Mick. I now have another problem with sending messages back to the phone, when I run: smsq -o0198339100 -q101 --mttx-channel sip/phone1 --ud test You need to run smsq as the Asterisk user, or else the file is created with permissions that Asterisk can't read and/or move. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
I've never seen this happen, and we run almost 300 calls through Asterisk with around 20 agents every day. How are you performing your attended transfer? Step-by-step. On 6/12/06, aston martin [EMAIL PROTECTED] wrote: It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active. Does anyone know any workarounds for this problem? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - SOLVED - Trouble getting SMS working
Mick, as good? How to configure sms with asterisk? Best Regards Josué 2006/6/12, Mick [EMAIL PROTECTED]: All working great now!Cheers.Mick. I now have another problem with sending messages back to the phone, when I run: smsq -o0198339100 -q101--mttx-channel sip/phone1 --ud test You need to run smsq as the Asterisk user, or else the file is created with permissions that Asterisk can't read and/or move. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' state Matt [EMAIL PROTECTED] wrote: I've never seen this happen, and we run almost 300 calls throughAsterisk with around 20 agents every day. How are you performingyour attended transfer? Step-by-step.On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote: It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active. Does anyone know any workarounds for this problem? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Single agent multiple queues....
Hi, I have several agents, who all log into multiple queues. What I want to happen (but doesn't seem to be) is: Agent 5 is logged into queues 1,2,3 Agent 4 is logged into queues 1,3 A call comes into queue 1, and goes to agent 5. Agent 5 answers the call and finishes it. A call comes into queue 3. I want this call to go to Agent 4, as opposed to going to agent 5 (which is what it is doing now). If a call comes into Queue 1, it WILL route to Agent 4. It seems that it only does round robin with memory per QUEUE and not per agent. Will changing to least-recently-called change this? Is there anyway to link the queues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - SOLVED - Trouble getting SMS working
Hi Josué,
These pages will give you some good info.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Sms
http://www.automated.it/asterisk/sms.html
I don't have the send from asterisk part ready yet, but to receive SMS
from a phone, in your extensions.conf you will need something like
this.
; SMS Message Center
SMSC=0198339100
; Receiving messages FROM a phone:
[smsphone]
exten = ${SMSC},1,Goto(smsmorx,s,1)
[smsmorx]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,SMS(${CALLERIDNUM},sa)
exten = s,4,System(someapptohandlelocalsms ${EXTEN})
exten = s,5,Hangup
I had a lot of problems with certain versions of app_sms.
What is working now for me is Asterisk 1.2.9.1, but with
app_sms.c from Asterisk 1.2.7.1.
Get it from here
http://svn.digium.com/view/asterisk/tags/1.2.7.1/apps/app_sms.c?rev=19815
Hope this helps.
Mick.
On Mon, 12 Jun 2006 08:26:38 -0300
Josué Conti [EMAIL PROTECTED] wrote:
Mick, as good?
How to configure sms with asterisk?
Best Regards
Josué
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Re: [Asterisk-Users] Attended transfer and queue
On 6/12/06, aston martin [EMAIL PROTECTED] wrote: 1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' state Is the agent's device you're dialing on the same server that app_queue is operating on? I've seen this happen when it is not on the same server, and there's a reason behind why it doesn't work at the present time in that configuration. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Single agent multiple queues....
On 6/12/06, Matt [EMAIL PROTECTED] wrote: Hi, I have several agents, who all log into multiple queues. What I want to happen (but doesn't seem to be) is: Agent 5 is logged into queues 1,2,3 Agent 4 is logged into queues 1,3 A call comes into queue 1, and goes to agent 5. Agent 5 answers the call and finishes it. A call comes into queue 3. I want this call to go to Agent 4, as opposed to going to agent 5 (which is what it is doing now). If a call comes into Queue 1, it WILL route to Agent 4. It seems that it only does round robin with memory per QUEUE and not per agent. Will changing to least-recently-called change this? Is there anyway to link the queues? No. chan_agent is nothing more than another device to app_queue which is the one making decisions about the ring strategy, and at this point, those decisions are made on a per-queue basis, not globally. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hitting * in a queue call hangs up?
Can anyone explain why when I hit * (as in *2 to transfer) a call that has come to me in a queue asterisk disconnects the call? All I have to do is hit * and the call drops. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - SOLVED - Trouble getting SMS working
Hi Mick. Its information will very help and I thank its attention, if I will be able help in some thing you can ask.
Regards
Josué
2006/6/12, Mick [EMAIL PROTECTED]:
Hi Josué,These pages will give you some good info.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Smshttp://www.automated.it/asterisk/sms.htmlI don't have the send from asterisk part ready yet, but to receive SMS
from a phone, in your extensions.conf you will need something likethis.; SMS Message CenterSMSC=0198339100; Receiving messages FROM a phone:[smsphone]exten = ${SMSC},1,Goto(smsmorx,s,1)
[smsmorx]exten = s,1,Answerexten = s,2,Wait(1)exten = s,3,SMS(${CALLERIDNUM},sa)exten = s,4,System(someapptohandlelocalsms ${EXTEN})exten = s,5,HangupI had a lot of problems with certain versions of app_sms.
What is working now for me is Asterisk 1.2.9.1, but withapp_sms.c from Asterisk 1.2.7.1.Get it from here
http://svn.digium.com/view/asterisk/tags/1.2.7.1/apps/app_sms.c?rev=19815Hope this helps.Mick.On Mon, 12 Jun 2006 08:26:38 -0300Josué Conti
[EMAIL PROTECTED] wrote: Mick, as good? How to configure sms with asterisk? Best Regards Josué___--Bandwidth and Colocation provided by
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Re: [Asterisk-Users] Attended transfer and queue
AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. On 6/12/06, aston martin [EMAIL PROTECTED] wrote: 1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' state Matt [EMAIL PROTECTED] wrote: I've never seen this happen, and we run almost 300 calls through Asterisk with around 20 agents every day. How are you performing your attended transfer? Step-by-step. On 6/12/06, aston martin wrote: It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active. Does anyone know any workarounds for this problem? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
I think it is. Agent's device is actually a softphone, registeredas one of the clients defined in sip.conf.Would it help if I posted the configs?BJ Weschke [EMAIL PROTECTED] wrote: On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote: 1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' stateIs the agent's device you're dialing on the same server thatapp_queue is operating on? I've seen this happen when it is not on thesame server, and there's a reason behind why it doesn't work at thepresent time in that configuration.-- Bird's The Word Technologies, Inc.http://www.btwtech.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 6/12/06, Matt [EMAIL PROTECTED] wrote: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. On 6/12/06, aston martin [EMAIL PROTECTED] wrote: 1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' state With 1.2.X, the * was hardcoded into chan_agent to drop the call. With /trunk and 1.4, this is now a configurable option. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hitting * in a queue call hangs up?
On 6/12/06, Matt [EMAIL PROTECTED] wrote: Can anyone explain why when I hit * (as in *2 to transfer) a call that has come to me in a queue asterisk disconnects the call? All I have to do is hit * and the call drops. This was a hardcoded feature in Asterisk 1.2.X versions. It's now an optional feature in /trunk and will be going forward. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] JIAX status
Scott, Could you point me some step-by-step instructions? I hadn't figure out what I'm doing wrong. I started over several times and did not find where I lost it. Many thanks in advance. On 6/12/06, Scott Gifford [EMAIL PROTECTED] wrote: Rubens Zupelli Filho [EMAIL PROTECTED] writes: You are compiling in Linux or Windows? Both. It works on Linux, but not yet on Windows. The package the java compiler is not founding is: net.sourceforge.iaxclient.jni That's part of the source package; probably the classpath just needs to be tweaked. ---Scott. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rubens Zupelli Filho [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 06/12/06 20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 6/12/06, BJ Weschke [EMAIL PROTECTED] wrote: On 6/12/06, Matt [EMAIL PROTECTED] wrote: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. On 6/12/06, aston martin [EMAIL PROTECTED] wrote: 1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' state With 1.2.X, the * was hardcoded into chan_agent to drop the call. With /trunk and 1.4, this is now a configurable option. Uhhh who's bright idea was that? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
use an fxo interface and 600ohm input on amp On Jun 11, 2006, at 9:53 AM, Thomas Kenyon wrote: Doug Lytle wrote: Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is Does the connection use 2 screws for analog inputs? Yup. If this is the case, you could get a cheap Grand Stream BT102 and pull the speaker leads off and connect it to that box. The GS can be setup to auto answer. Doug I'm going to try the suggestion in the Bat Phone thread above, bringing one of the PAP2s out of retirement. Wish me luck. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 6/12/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 06/12/06 20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied. Ahhh I always love a good upgrade :P I'd just as soon change the transfer to #2, however my aastra phones don't seem to let me use #270 in my softkeys.. as soon as it hits # it just dies and stoppes... anyone know a way around this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] JAMAICA DID'S - 1-876
JAMAICA DID'S - 1-876 NOW ACTIVE ON www.didx.org Cheers, Oliver VermeulenWorld Venture Group Telecom Office: +(40)21-569-4700Office2: +(40)31-860-0030Fax: +(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK DID:+(44)870-478-8896msn: [EMAIL PROTECTED]http://www.wvg-tele.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far..Dinesh Nair [EMAIL PROTECTED] wrote: On 06/12/06 20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect.take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied.-- Regards, /\_/\ "All dogs go to heaven."[EMAIL PROTECTED] (0 0) http://www.openmalaysiablog.com/+==oOO--(_)--OOo==+| for a in past present future; do || for b in clients employers associates relatives neighbours pets; do || echo "The opinions here in no way reflect the opinions of my $a $b." || done; done |+=+___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cell gateway for T-Mobile US??
Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the asterisk side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... Jerry Jones wrote: use an fxo interface and 600ohm input on amp On Jun 11, 2006, at 9:53 AM, Thomas Kenyon wrote: Doug Lytle wrote: Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is Does the connection use 2 screws for analog inputs? Yup. If this is the case, you could get a cheap Grand Stream BT102 and pull the speaker leads off and connect it to that box. The GS can be setup to auto answer. Doug I'm going to try the suggestion in the Bat Phone thread above, bringing one of the PAP2s out of retirement. Wish me luck. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
On 6/12/06, aston martin [EMAIL PROTECTED] wrote: Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far.. That would depend on what's causing this to happen. If you open up a bug at bugs.digium.com with the configs and trace/logs, we'll take a look at what's going on. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
What version of Asterisk are you running, that you are able to dial *2 and the * isn't hanging up like it is for me? On 6/12/06, aston martin [EMAIL PROTECTED] wrote: Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far.. Dinesh Nair [EMAIL PROTECTED] wrote: On 06/12/06 20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED] (0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Jerry Jones wrote: use an fxo interface and 600ohm input on amp On Jun 11, 2006, at 9:53 AM, Thomas Kenyon wrote: Doug Lytle wrote: Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is Does the connection use 2 screws for analog inputs? Yup. If this is the case, you could get a cheap Grand Stream BT102 and pull the speaker leads off and connect it to that box. The GS can be setup to auto answer. Doug I'm going to try the suggestion in the Bat Phone thread above, bringing one of the PAP2s out of retirement. Wish me luck. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail issue
Hello, I have created a voicemail with succes for user victor. But a second vocemail for user julian, asterisk claims that the voicemail for user 'lian' is not configured, why asterisk is getting rid of the first 2 chars of 'julian'. For the moment I have created user aajulian in both extensions.cfg and voicemail.cfg and with that tricks it works well, but i would like to know why i cannot use the username julian. Thanks Victor ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spa3102 vs spa3000 differences?
Anyone know what the differences are between the spa3000 and spa3102 other then packaging? R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cell gateway for T-Mobile US??
Typically yes, as long as you can get power for them compatible with ours. Tmobile is GSM. Well only GSM. They don't do anything else. You can check the WIKI I have found a few smaller ones that will probably work but don't remember what they are except that I found them there. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz, Steven Sent: Monday, June 12, 2006 9:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cell gateway for T-Mobile US?? Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the asterisk side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended transfer and queue
I'm using 1.2.9.1, and it isn't hanging up... (at leastTHAT seems to be okay, hehe). I wonder how nobody else came across this before, cause I'm not using queues pretty much by the bookthe only thing is that agents stay busy after transfer.Here are part of my configs, if somebody gets any ideas, what could be causing that:That's the sip client (defined in sip.conf):[Agent001]username=Agent001secret=Agent001type=friendhost=dynamiccontext=from-sipdisallow=allallow=alawallow=ulaw- That's the extension with the queue:exten = _995,1,Answer()exten = _995,2,LookupBlacklist(j)exten = _995,3,Set(MONITOR_FILENAME=${CALLERIDNUM}_${UNIQUEID}_${EXTEN}_wav128)exten
= _995,4,Queue(MainQueue|tT|||14400)- That's the extension where agent gets actually called:exten = _0XXX.,1,Dial(SIP/Agent${EXTEN:1:3},8,tTj) And that's pretty much all. Matt [EMAIL PROTECTED] wrote: What version of Asterisk are you running, that you are able to dial *2and the * isn't hanging up like it is for me?On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote: Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far.. Dinesh Nair <[EMAIL PROTECTED]>wrote: On 06/12/06
20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect. take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED] (0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done |
+=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com
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Re: [Asterisk-Users] Attended transfer and queue
On 06/12/06 20:42 Dinesh Nair said the following: i would think that 1.2.9.1 would also have this patch applied. not it doesnt. my patch was only committed for trunk, though mantis does have the patch that works on 1.2.x as well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
John Novack wrote: Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa3102 vs spa3000 differences?
Rich Adamson [EMAIL PROTECTED] wrote: Anyone know what the differences are between the spa3000 and spa3102 other then packaging? The 3102 includes a router (two RJ45s). -Darren -- Darren Nickerson Senior Sales Engineer Telephony Depot www.telephonydepot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon Europe - Only 1 Week Away
Remember that AstriCon Europe kicks off in only a week with the opening in Berlin. Other events follow in Paris and London. Join us and get to know the Asterisk community in person. We hope to see you there. For more info or to register: http://www.astricon.net Thanks, Steve -- Steven Sokol AstriCon 2006: http://www.astricon.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell gateway for T-Mobile US??
In Europe, the 900 and 1800MHz bands are used for GSM. In the USA, the 800 (or 850 as some call it) and 1900MHz bands are used for GSM as well as other protocols. T-Mobile USA uses 1900MHz GSM exclusively, although they do have a few territories in which GSM 800/850 roaming is allowed. So if you want to connect a GSM gateway to T-Mobile, it must support the 1900MHz band, and if you want to use it in some of T-Mobile's roaming areas, it should also support the 800/850MHz band. -Rusty On 6/12/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the asterisk side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP auth failed wrong pw but pw is correct
Hi We've got an Asterisk 1.2.0 (planning to upgrade when I can) which is having trouble registering another Asterisk system as a client. We have the client in a realtime DB, our client has us configured as a friend and also has a register = username:[EMAIL PROTECTED]/username line in his sip.conf. When he starts his system he gets an auth failed log line and we get: Jun 12 14:42:21 NOTICE[1548]: chan_sip.c:10817 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '81.187.x.x' - Wrong password The client has the right username and password. One difference I've noticed is that the log error says Registration from 'sip:[EMAIL PROTECTED]' failed for '81.187.x.x' -- every other registration line quotes only the SIP account name, not the whole SIP URI. This problem does look similar to bug #5103 but the fix should be incorporated in Asterisk 1.2.0, I think. SIP trace: Sending to 81.187.x.x : 5060 (non-NAT) Transmitting (NAT) to 81.187.x.x:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.187.x.x:5060;branch=z9hG4bK26bc6cc8;received=81.187.x.x;rport=5060 From: sip:[EMAIL PROTECTED];tag=as48322a50 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 81.187.x.x:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 81.187.x.x:5060;branch=z9hG4bK26bc6cc8;received=81.187.x.x;rport=5060 From: sip:[EMAIL PROTECTED];tag=as48322a50 To: sip:[EMAIL PROTECTED];tag=as7592c16c Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=63563ef1 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms sip2*CLI -- SIP read from 81.187.x.x:5060: REGISTER sip:fdqn.our.host SIP/2.0 Via: SIP/2.0/UDP 81.187.x.x:5060;branch=z9hG4bK06e5ad1f;rport From: sip:[EMAIL PROTECTED];tag=as3c9883b7 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=username, realm=asterisk, algorithm=MD5, uri=sip:fdqn.our.host, nonce=63563ef1, response=d6daaafb77c11357bda9b912b8ce6e02, opaque= Expires: 18000 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 81.187.x.x : 5060 (NAT) Transmitting (NAT) to 81.187.x.x:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.187.x.x:5060;branch=z9hG4bK06e5ad1f;received=81.187.x.x;rport=5060 From: sip:[EMAIL PROTECTED];tag=as3c9883b7 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 81.187.x.x:5060: SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 81.187.x.x:5060;branch=z9hG4bK06e5ad1f;received=81.187.x.x;rport=5060 From: sip:[EMAIL PROTECTED];tag=as3c9883b7 To: sip:[EMAIL PROTECTED];tag=as7592c16c Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Any ideas? Cheers, Mark This message and any attachment are confidential and may be privileged or otherwise protected from disclosure. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment(s) from your system and do not disclose its contents to any third parties. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spa3102 vs spa3000 differences?
Anyone know what the differences are between the spa3000 and spa3102 other then packaging? The SPA3102 has a built-in router (LAN + WAN), T.38 on the FXS port and can handle dual G.729 sessions. Nabeel Jafferali www.voipdepot.ca ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Thomas Kenyon wrote: John Novack wrote: Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). Won't you still need to maintain the loop current to make the PAP2 look like the port is off-hook? (FXS BTW) I would think the impedance matching xformer falls int the need to category. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Thomas Kenyon wrote: John Novack wrote: Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). Won't you still need to maintain the loop current to make the PAP2 look like the port is off-hook? (FXS BTW) I would think the impedance matching xformer falls int the need to category. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa3102 vs spa3000 differences?
Nabeel Jafferali wrote: Anyone know what the differences are between the spa3000 and spa3102 other then packaging? The SPA3102 has a built-in router (LAN + WAN), T.38 on the FXS port and can handle dual G.729 sessions. Has there been any improvements in the echo cancellation functions associated with the fxo port? R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Stderr
Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Stderr
Hi Douglas, Try this: open(STDERR, /etc/asterisk/agi-bin/errors.txt) Fred - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 12, 2006 11:32 Subject: [Asterisk-Users] AGI Stderr Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa3102 vs spa3000 differences?
On the spa3000 the firmware version seems to greatly effect the echo. Mine came with 2.x firmware which was fine, I upgraded to (latest) 3.1.10d and it was horrible. I then downgraded to 3.1.3a and it was OK, but not as good a 2.x version. The reason I upgraded in the first place was to allow FXO calls to be passed but not answered, which was not a 2.x feature. It seems that any firmware is usable on any hardware as my hardware is 2.x. I wonder if 3102 firmware could be used on the 3000. Is the size the same? I guess you would have to be willing to make a brick to find out! Doug On Mon, 12 Jun 2006, Rich Adamson wrote: Nabeel Jafferali wrote: Anyone know what the differences are between the spa3000 and spa3102 other then packaging? The SPA3102 has a built-in router (LAN + WAN), T.38 on the FXS port and can handle dual G.729 sessions. Has there been any improvements in the echo cancellation functions associated with the fxo port? R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Qsig
Hello, can you tell me, if called id name display is working in pbx-asterisk interworking using Q.SIG protocol? (I have siemens hipath pbx). thx PJ Michael Konietzny wrote: Hello Josué, yes i currently only switched switchtype in zapata.conf to the value qsig. The only real PRI feature i've found out is the PRI_CAUSE variable set on Hangup(). Greetings, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa3102 vs spa3000 differences?
On 6/12/06, Doug Crompton [EMAIL PROTECTED] wrote: It seems that any firmware is usable on any hardware as my hardware is 2.x. I wonder if 3102 firmware could be used on the 3000. Is the size the same? I guess you would have to be willing to make a brick to find out! I have not tried this, but on an spa2000, the firmware updater simply made no changes when I tried to install some unsupported firmware. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Stderr
Oh yeah, I also won't get time/date stamps if I redirect stderr to a file like that -Original Message- From: Douglas Garstang Sent: Monday, June 12, 2006 8:51 AM To: 'Frederic Jean' Subject: RE: [Asterisk-Users] AGI Stderr Frederic, Thanks, but that's not the best approach. I am sending all debug from my AGI script to syslog. I'd like runtime errors to go to Asterisk so that it can log them to a file. If I don't, I'll have files in three places instead of two. (syslog, errors.txt and /var/log/asterisk/*) Doug. -Original Message- From: Frederic Jean [mailto:[EMAIL PROTECTED] Sent: Monday, June 12, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI Stderr Hi Douglas, Try this: open(STDERR, /etc/asterisk/agi-bin/errors.txt) Fred - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 12, 2006 11:32 Subject: [Asterisk-Users] AGI Stderr Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio cuts out
Hey All,I've been experiencing a problem for a bit. During a call to the PSTN, audio will cut out for 2-5 seconds. It's completely random and may or may not happen during a call.Our setup is 79XX phones - asterisk - 2811 router - PRI to the PSTN. Everything is talking SIP. The asterisk box is a dual core system. /proc/interrupts looks like: cat /proc/interrupts CPU0 CPU1 0: 733669449 732813122 IO-APIC-edge timer 8: 1 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 6598410 6589174 IO-APIC-edge ide0169: 0 0 IO-APIC-level uhci_hcd185: 0 0 IO-APIC-level ehci_hcd, uhci_hcd193: 0 0 IO-APIC-level uhci_hcd 201: 0 0 IO-APIC-level uhci_hcd209: 11404158 10762030 IO-APIC-level 3w-9xxx225: 100440701 136 PCI-MSI eth0233: 14 10512166 PCI-MSI eth1NMI: 0 0 LOC: 1466464719 1466464718 ERR: 0MIS: 0Can-Reinvite is enabled, but I do have it configured to allow call recording on outbound calls, so I think the audio streams all go through asterisk. There are no G.729 licenses involved and everything should be talking G.711. Oh, and this is an 1.2.7.1 install. ztdummy is loaded.Does anyone have any insite into this problem?Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Qsig
Hello Pavel. I still did not make a test, but I will be making one upgrade in asterisk to apply qsig HiPath 4000, thus that to make the testssend an email for the list. Which the version of its HiPath? Best Regards Josué 2006/6/12, Pavel Jezek [EMAIL PROTECTED]: Hello, can you tell me, if called id name display is working inpbx-asterisk interworking using Q.SIG protocol? (I have siemens hipath pbx). thxPJMichael Konietzny wrote: Hello Josué, yes i currently only switched switchtype in zapata.conf to the value qsig. The only real PRI feature i've found out is the PRI_CAUSE variable set on Hangup(). Greetings, Michael___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem dialing out thru sip - using isdn on internal
hi,
i've a wirded problem, i try to dial out, using this dialplan
[default]
exten = _*7.,1,Macro(anrufextern-sip,${EXTEN:2})
[macro-anrufextern-sip]
exten = s,1,SetCallerID(SIP-ID)
exten = s,n,Dial(SIP/${ARG1}sip-out)
exten = s,n,Hangup()
when i use my analog telephone, everything is okay:
- Starting simple switch on 'Zap/3-1'
-- Executing Macro(Zap/3-1, anrufextern-sip|9199125) in new stack
-- Executing SetCallerID(Zap/3-1, SIP-ID) in new stack
-- Executing Dial(Zap/3-1, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sip-out-0fe9 is ringing
-- SIP/sip-out-0fe9 is ringing
== Spawn extension (macro-anrufextern-sip, s, 2) exited non-zero on
'Zap/3-1' in macro 'anrufextern-sip'
== Spawn extension (macro-anrufextern-sip, s, 2) exited non-zero on
'Zap/3-1'
-- Hungup 'Zap/3-1'
but when i dial from my isdn phone, it dials as soon as it gets the
first digit of the phone number
and does not wait for the 199125
-- Executing Macro(mISDN/2-u12, anrufextern-sip|9) in new stack
-- Executing SetCallerID(mISDN/2-u12, SIP-ID) in new stack
-- Executing Dial(mISDN/2-u12, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sip-out-5d40 is circuit-busy
any ideas?
are there any switches in the misdn.conf providing this?
using :
misdn 0.3.1-rc11
asterisk 1.2.7
___
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Re: [Asterisk-Users] AGI Stderr
, never tried it with asterisk but you could redirect STDERR to STDOUT and see how you can capture this guy afterward... open STDERR, STDOUT; just a thought - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 12, 2006 11:52 Subject: RE: [Asterisk-Users] AGI Stderr Oh yeah, I also won't get time/date stamps if I redirect stderr to a file like that -Original Message- From: Douglas Garstang Sent: Monday, June 12, 2006 8:51 AM To: 'Frederic Jean' Subject: RE: [Asterisk-Users] AGI Stderr Frederic, Thanks, but that's not the best approach. I am sending all debug from my AGI script to syslog. I'd like runtime errors to go to Asterisk so that it can log them to a file. If I don't, I'll have files in three places instead of two. (syslog, errors.txt and /var/log/asterisk/*) Doug. -Original Message- From: Frederic Jean [mailto:[EMAIL PROTECTED] Sent: Monday, June 12, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI Stderr Hi Douglas, Try this: open(STDERR, /etc/asterisk/agi-bin/errors.txt) Fred - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 12, 2006 11:32 Subject: [Asterisk-Users] AGI Stderr Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Presentation + Asterisk Realtime doubts
Hello everyone, I'm Andrea, and I've started working with Asterisk a couple of weeks ago, so I'm still a newbie. :) I was reading about Asterisk Realtime, and I was wondering if I can mix Static realtime and Real realtime configuration. For instance: can I have a Static Realtime extensions.conf and use Real Realtime sippeers and sipusers? Moreover, is there an easy way to switch between Static realtime and Real realtime mode for the realtime families? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Bob Chiodini wrote: Thomas Kenyon wrote: Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). Won't you still need to maintain the loop current to make the PAP2 look like the port is off-hook? (FXS BTW) Err, yeah, the PAP2 only has FXS ports. I would think the impedance matching xformer falls int the need to category. It would do, if they were different impedances. This isn't a normal audio amp, it expects to be plugged into an extension port of an analogue phone system (which sees it as permanently off-hook). I'm not there to test any of this, but I did plug it into an FXS port of the AG-468 that's there to see if it sounded okay, and the whole place rumbled with the configured dialtone :-) So hopefully the same will apply with a PAP2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Unless you pruchae an exteral zone page adapter which accepts FXS coonections, use an ata with fxo connection I have never had to 'add' loop current, although I have not used all fxo adapters. I can verify Adit and other channel bank fxo connections do not require any voltage, the just give the analog audio out On Jun 12, 2006, at 9:24 AM, Bob Chiodini wrote: Thomas Kenyon wrote: John Novack wrote: Bob Chiodini wrote: You might put 600 ohm/600 ohm matching transformer to isolate the port and the amp. Should also maintain loop current if needed. Bob... FXO ports do not generate loop current, they detect loop current from the Central office. Think of an FXO as a controllable switch and matching transformer. An FXO port will give Asterisk/Zaptel a red alarm without loop current, so that probably won't work for a PA system that doesn't supply battery. What's wrong with using the (usually) unused sound card built into many machines? John Novack Mostly it uses the wrong impedance, and I know I can probably get an impedance matching transformer, but I'm not allowed to spend any money that I don't need to. (Otherwise I'd have replaced the amp in the first place with one that didn't fall from the arc). Won't you still need to maintain the loop current to make the PAP2 look like the port is off-hook? (FXS BTW) I would think the impedance matching xformer falls int the need to category. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CallerID name inbound from PRI
XO fixed my caller ID name.
I am using FreePBX and I can include a wait to my custom extensions.
Is there a way to add a wait to the whole PRI?
I assume that if I set immediate to yes, I can then have a s extension
do the wait, but how would it get from the s to the DID extension?
(also, I would rather not answer every call)
Is there a magic spot in Free PBX's configs to add the wait for all
calls on that PRI, or do I need to alter the FreePBX code to add it when
creating the conf. Files?
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
TESCO Group Companies
Fax. 248-836-5101
www.TESCOGroup.com
Board member of
www.glimasoutheast.org
Steven [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Thanks for the info.
I went to add the Wait(2), but am unsure where to do it.
My context is from-pstn.
My [from-pstn] is:
[from-pstn]
exten = s,1,NoOp(${TIMESTAMP} PRI call in) ;I tried adding this
to see if s is used, but lothing was logged.
include = from-pstn-custom ; create this context
in extensions_custom.conf to include customizations
include = ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
My from-pstn-custom is non-existent and my ext-did is just an
include for ext-local, which is my inside extensions.
If I understand you correctly, I need the wait before I pick up the
line.
If I change the span to immediate=yes, I can use the s extension,
but It would also answer the line early.
I am drawing a blank where to put the wait.
Please advise.
--
--
Steven
http://www.glimasoutheast.org
Alexander Lopez [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
There is nothing you really need 'to do' if your PRI is working
already,
If you are able to receive and make calls your D-Channel is
functioning
properly. In the case of CallerID, some telcos provide this extra
function via the FACILITY messages instead of the SETUP messages, If
that is the case, you will get no Name but you will get a number. IT
simply means that Asterisk answered the call with the SETUP message
but
was unable to read in the CALLERID Name to pass on to your devices
because it comes later on in the call via the FACILITY.
Add a Wait(2) before you answer the call for your PRI, see if that
helps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Monday, April 10, 2006 8:57 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] callerid name inboune from PRI
I switched PRI vendors recently, and one of my questions was
do you provide caller ID name in addition to number?
ATT Local did not, But XO communications said they did.
Before I call to complain, is there an setting to turn this
on in asterisk?
I want to make sure that I have my side covered before I call XO.
My current zaptel.conf is:
context=from-pstn
switchtype=national
pridialplan=unknown
prilocaldialplan=unknown
priindication = outofband
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=no
group=0
callgroup=1
pickupgroup=1
accountcode=I
musiconhold=default
channel = 1-23
--
--
Steven
http://www.glimasoutheast.org
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Re: [Asterisk-Users] JIAX status
Rubens Zupelli Filho [EMAIL PROTECTED] writes: Scott, Could you point me some step-by-step instructions? I hadn't figure out what I'm doing wrong. I started over several times and did not find where I lost it. I just did configure and make, then fixed all the problems that came up. I don't have any step-by-step instructions, and I don't have my small fixes into any kind of publishable form. Scott. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO registration and VegaStream
I do have extension 13 in sip.conf but I still get Destroying call on all incoming calls coming from VegaStream. [13] type=user dtmfmode=inband disallow=all context=from-trunk allow=alaw -- SIP read from 192.168.0.5:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK-vega1-000A-0001-002A-C9DAAC64 From: FJLine2 sip:[EMAIL PROTECTED];tag=-002B-F7607240 To: sip:[EMAIL PROTECTED];tag=as572a6b57 Max-Forwards: 70 Call-ID: [EMAIL PROTECTED] CSeq: 6433460 ACK Contact: sip:[EMAIL PROTECTED]:5060;maddr=192.168.0.5 User-Agent: VEGAPOTS/09.02.07xS008 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Doyle Sent: Monday, June 12, 2006 1:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FXO registration and VegaStream Hi Issac, Ok, here goes :) Again, my disclaimer-- I'm pretty new to Asterisk, so I'm sure half of this is not needed or potentially even misconfigured. You will even see some lines commented out, since I wanted to test if they were needed--they weren't. I'm hoping to clean everything up and put it on the wiki -- hopefully next week or two. Also, these are from Asterisk @ Home, so there might be some changes needed for your setup. * Sip.conf - context line may differ from [EMAIL PROTECTED] Defaults * [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf * * Sip_additional.conf - * I haven't tested DTMF on incoming calls-- you may have to * change dtmfmode to inband (rfc2833 didn't work for the outgoing * calls). Also, the context may need to be changed for security? * I only have an entry for 01 since I am testing with 1 line only * ... snip ... [01] ;most lines added by [EMAIL PROTECTED], may not be necessary (i.e. mailbox) username=01 type=friend secret=...my vega's password for line 1... (see POTS in Vega's web config) record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 01 ... snip ... ; commented out, doesn't seem to be needed ;[vega] ;type=user ;dtmfmode=inband ;disallow=all ;context=from-pstn ;allow=ulaw [vega-gw] type=peer host=192.168.1.30 ; my vega's IP address dtmfmode=inband ;DTMF doesn't work with rfc2833, unfortunately disallow=all ;context=from-internal ; commenting out, makes context default to from-sip-external? allow=ulaw ;only allow ulaw * * extensions_additional.conf - dials extension 106 on incoming * call. I think there's some special [EMAIL PROTECTED] magic happening in the * macro to dial 106. You could just have something like Dial() * happen here. * * After adding the 06 extension, that is when incoming calls * start going through. * * You could also use the s extension somehow, as Mike showed us * (I need to read up a little!! :) ) * exten = 06,1,Macro(exten-vm,novm,06) exten = 06,hint,SIP/106 * * Configuration Change Report from the Vegastream * (shows changes from factory settings) * Report on configuration changes (verbose) Configuration changes: Key: CU: Changed from factory and unsaved. C-: Changed from factory and saved. -U: Not changed but unsaved. [call_control.timers.1] T301_timeout=90 T301_cause=18 [dsp.g711Alaw64k] VADU_threshold=0 VP_FIFO_max_delay=160 VP_FIFO_nom_delay=60 echo_tail_size=16 idle_noise_level=-7000 packet_time_max=30 packet_time_min=10 packet_time_step=10 rx_gain=0 tx_gain=0 [dsp.g711Alaw64k.data] EC_enable=disable [dsp.g711Alaw64k.voice] EC_enable=enable [dsp.g711Ulaw64k] ;I'm only using Ulaw, so this is the only codec set up VADU_threshold=0 C- VP_FIFO_max_delay=60 *factory=160 C- VP_FIFO_nom_delay=10 ; I figured reducing this is ok (Asterisk - vega is on a LAN), and might reduce delay? *factory=40 C- echo_tail_size=8 ; EC trains much faster @ 8ms tail for me (we are close to CO) *factory=16 idle_noise_level=-7000 C- packet_time_max=20 ; Asterisk requires 20ms packets for ULAW
Re: [Asterisk-Users] Attended transfer and queue
On 06/12/06 21:11 Matt said the following: What version of Asterisk are you running, that you are able to dial *2 and the * isn't hanging up like it is for me? because i wrote and applied the patch ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: TTS from MySQL
Hi all, I need to simply use Asterisk to receive incoming calls in an IVR manner. It should authenticate users and read data from MySQL table that match their ID through Text-to-speech. I already have Asterisk 2.6 ([EMAIL PROTECTED]). I understand that I need to use Festival and AGI but do not know what to do exactly. Any help is appreciated. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX DID channels as incoming hunt group?
Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the carrier or provider, but I want to make sure: if they are IAX channels, is there any way to do this in Asterisk on the receiving end? Shouldn't a provider offering IAX DID be able to do this for me before the calls are sent to my Asterisk server? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem dialing out thru sip - using isdn on internal
got following hint from c.richter from beronet support team
exten = _8.,1,waitfordigits(4000)
exten = _8.,n,Macro(anrufextern-sip,${EXTEN:1})
exten = _9.,1,waitfordigits(4000)
exten = _9.,n,Macro(anrufextern-analog,${EXTEN:1})
now it gets all digits
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[Asterisk-Users] RAGI + Sphinx + Festival
Hi All, Has anybody ever tried to use Sphinx and Festival from Ruby AGI scripts (Ruby on Rails and AGI) ? Please share your experience or even samples of code - that would be great. Thank you, Andrei (MPI) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hitting * in a queue call hangs up?
- BJ Weschke [EMAIL PROTECTED] wrote: This was a hardcoded feature in Asterisk 1.2.X versions. It's now an optional feature in /trunk and will be going forward. And this is only true for queue members that are chan_agent agents. If you don't use chan_agent, you won't see this behavior either. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
Stephen Bosch wrote: Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the carrier or provider, but I want to make sure: if they are IAX channels, is there any way to do this in Asterisk on the receiving end? Shouldn't a provider offering IAX DID be able to do this for me before the calls are sent to my Asterisk server? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In the VoIP world technologically there's no reason why you would need an incoming hunt group as you say. As a call comes in, the provider would forward it to you - there's no actual group of channels... just your account information. Now - some providers may limit how many you can actually have up simultaneously but that's a feature of their system. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX DID channels as incoming hunt group?
You just do pattern matching in your dialplan: [from-myIAXprovider] exten = 55512XX,1,Dial(SIP/reception,40,T) exten = 55512XX,2,Voicemail() So anything coming in with a dialled extension of 55512XX will pattern-match to the above lines. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Monday, June 12, 2006 10:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX DID channels as incoming hunt group? Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the carrier or provider, but I want to make sure: if they are IAX channels, is there any way to do this in Asterisk on the receiving end? Shouldn't a provider offering IAX DID be able to do this for me before the calls are sent to my Asterisk server? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 passthrough/middleman
That sounds exactly like what I want to do. I've don't have a PRI line (although I'm going to press for getting one soon), but for now I would just like a couple of pointers in getting Asterisk's dial plan set up to just pass the calls from one T1 to another. Thanks a million in advance. Mimmus wrote: I used this approach to gradually migrate from a legacy Alcatel PBX: PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel PBX At first, Asterisk did nothing, only passing calls to/from Alcatel. Then I started to use a bunch of SIP phones directly connected to Asterisk. Now I have the great part of extensions as SIP phones and the old PBX is working as a channel bank only for a few of analog devices. Configuring the dialplan to do this dirty job is not difficult but now I'm not able to help you because it's saturday evening and I'm at home! Re-try next Monday. DV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bell Sent: Friday, June 09, 2006 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] T1 passthrough/middleman Is it possible to act as a middle man on a T1 line? My installation currently has an aging Inter-Tel Axxess box with a T1 coming in (16 in, 8 out). Rather than adding and replacing phones and cards as they die, I would like to slowly migrate to a asterisk SIP installation. I want to take the incoming T1 line, use any available outgoing lines for outgoing SIP, intercept any incoming lines and either send them off to a SIP line or pass them through to other T1 line (going to the Axxess box), and finally take in outgoing calls from the Inter-Tel box and either send them to SIP or send them to the outside T1 line. How will a dual T1 card be set up in this situation? Would it be easier to use an FXO channel bank (or card) and connect analog lines to the FXS analog lines on the Inter-Tel box? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
- Stephen Bosch [EMAIL PROTECTED] wrote: Shouldn't a provider offering IAX DID be able to do this for me before the calls are sent to my Asterisk server? This is completely unnecessary. A provider giving you IAX DID service can send you as many channels as they (and you) want, for a single incoming number. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio cuts out
Gary, I would check echo cancelling parameters first. I've seen this to happen with one of the zaptel echo cancellers. Try to change the default echo algorithm in zconfig.h, and recompile and install new zaptel. Also zapata.conf echo parameters may need to be changed either way. Andrei Gary Richardson wrote: Hey All, I've been experiencing a problem for a bit. During a call to the PSTN, audio will cut out for 2-5 seconds. It's completely random and may or may not happen during a call. Our setup is 79XX phones - asterisk - 2811 router - PRI to the PSTN. Everything is talking SIP. The asterisk box is a dual core system. /proc/interrupts looks like: cat /proc/interrupts CPU0 CPU1 0: 733669449 732813122IO-APIC-edge timer 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14:65984106589174IO-APIC-edge ide0 169: 0 0 IO-APIC-level uhci_hcd 185: 0 0 IO-APIC-level ehci_hcd, uhci_hcd 193: 0 0 IO-APIC-level uhci_hcd 201: 0 0 IO-APIC-level uhci_hcd 209: 11404158 10762030 IO-APIC-level 3w-9xxx 225: 100440701136 PCI-MSI eth0 233: 14 10512166 PCI-MSI eth1 NMI: 0 0 LOC: 1466464719 1466464718 ERR: 0 MIS: 0 Can-Reinvite is enabled, but I do have it configured to allow call recording on outbound calls, so I think the audio streams all go through asterisk. There are no G.729 licenses involved and everything should be talking G.711. Oh, and this is an 1.2.7.1 http://1.2.7.1 install. ztdummy is loaded. Does anyone have any insite into this problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] get value from DB directly
Hi, I want to know how I can get a value from a table. Say, I have a table sip_buddies for storing sip user account information. There is a field called 'accountcode' that I want to get its value in the dial plan. As I find that there is no direct way to get the value from the table. Does anyone can tell me how can I get its value in the dial plan? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
Joshua Colp wrote: In the VoIP world technologically there's no reason why you would need an incoming hunt group as you say. As a call comes in, the provider would forward it to you - there's no actual group of channels... just your account information. Now - some providers may limit how many you can actually have up simultaneously but that's a feature of their system. That's kinda what I was thinking. So -- to clarify that -- it's technically possible to have a single DID that allows multiple calls to be set up. The DID is just the line identifier, but we could have say three simultaneous calls, as long as the provider allows it -- correct? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
Colin Anderson wrote: You just do pattern matching in your dialplan: [from-myIAXprovider] exten = 55512XX,1,Dial(SIP/reception,40,T) exten = 55512XX,2,Voicemail() So anything coming in with a dialled extension of 55512XX will pattern-match to the above lines. That only works if the caller has dialled one of the other numbers. If we only publish one number (555 1212), what then? -s ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM Fax Problems
I am running into errors in faxes received by my * server that I route to fax machines on a channel bank. I have a channelized T-1 coming in and I take certain channels and transfer them to channels on my channel bank. The T-1 and channel bank are on a Sangoma A104D 4 port with echo canceling. The faxes tend to have problems with sections of the transmission getting garbled or lost. Does anyone have any information on why this might be happening or how to fix it? -- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) And zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callprogress=yes ; define channels context=incoming signalling=fxs_ks channel = 4 Pretty straightforward stuff -- a call comes in on the PSTN line, the Asterisk answers the call, then rings the extension. The caller hears a ring tone throughout the entire process. The rub is that Asterisk has, in reality, taken the PSTN line off hook. Not great if the caller is at a payphone. What if nobody answers the extension? The caller is out his money (50 cents in most of the US, 35 cents in Alberta and 25 cents in the rest of Canada ;) ) So I had the idea of doing things a bit differently, like so: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Dial(SIP/polycom) exten = s,2,Answer() This way, Asterisk dials the extension first, the idea being that when the SIP extension is answered, Asterisk answers the PSTN line and connects the channels. This did not have the expected result -- when I tried this, my SIP extension rang, but answering the extension did not result in Asterisk picking up the PSTN line. There is a way of doing this, isn't there? How can I make this work? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio cuts out
We're not using any zaptel hardware though. I didn't think the echo cancellers would be doing anything? We're digital and sip from end to end. Do I need to disable echo cancellation in some way? Thanks.On 6/12/06, Andrei (MPI) [EMAIL PROTECTED] wrote: Gary,I would check echo cancelling parameters first. I've seen this to happenwith one of the zaptel echo cancellers. Try to change the default echoalgorithm in zconfig.h,and recompile and install new zaptel. Also zapata.conf echo parameters may need to be changed either way.AndreiGary Richardson wrote: Hey All, I've been experiencing a problem for a bit. During a call to the PSTN, audio will cut out for 2-5 seconds. It's completely random and may or may not happen during a call. Our setup is 79XX phones - asterisk - 2811 router - PRI to the PSTN. Everything is talking SIP. The asterisk box is a dual core system. /proc/interrupts looks like: cat /proc/interruptsCPU0 CPU1 0:733669449732813122IO-APIC-edgetimer 8:10IO-APIC-edgertc 9:00 IO-APIC-levelacpi 14:65984106589174IO-APIC-edgeide0 169:00 IO-APIC-leveluhci_hcd 185:00 IO-APIC-levelehci_hcd, uhci_hcd 193:00 IO-APIC-leveluhci_hcd 201:00 IO-APIC-leveluhci_hcd 209: 11404158 10762030 IO-APIC-level3w-9xxx 225:100440701136 PCI-MSIeth0 233: 14 10512166 PCI-MSIeth1 NMI:00 LOC: 1466464719 1466464718 ERR:0 MIS:0 Can-Reinvite is enabled, but I do have it configured to allow call recording on outbound calls, so I think the audio streams all go through asterisk. There are no G.729 licenses involved and everything should be talking G.711. Oh, and this is an 1.2.7.1 http://1.2.7.1 install. ztdummy is loaded. Does anyone have any insite into this problem? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
Stephen Bosch wrote: So -- to clarify that -- it's technically possible to have a single DID that allows multiple calls to be set up. The DID is just the line identifier, but we could have say three simultaneous calls, as long as the provider allows it -- correct? You got it. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
Kevin P. Fleming wrote: - Stephen Bosch [EMAIL PROTECTED] wrote: Shouldn't a provider offering IAX DID be able to do this for me before the calls are sent to my Asterisk server? This is completely unnecessary. A provider giving you IAX DID service can send you as many channels as they (and you) want, for a single incoming number. Good news! Oh, the possibilities! Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX DID channels as incoming hunt group?
no from the Asterisk perspective it will work regardless of the number dialled as long as it matches the 55512XX pattern. As others have pointed out though, it's just easier to have a single DID and your provider allow multiple channels or instances of the same number to hit your box. it's hard sometimes to let go of concepts from the legacy PBX world when you try to integrate their equivalent in Asterisk. For example, I still enforce the dial 9 to get an outside line in my setups even though there is no technical reason for doing do. The reason is because people can't deal with dialling direct and it becomes a training issue. The other one that always makes me laugh is people depressing the hookswitch to flip between calls even though they have line indicator buttons right there, flashing on their phone. Fortunately Snom phones allow this functionality. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Monday, June 12, 2006 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX DID channels as incoming hunt group? Colin Anderson wrote: You just do pattern matching in your dialplan: [from-myIAXprovider] exten = 55512XX,1,Dial(SIP/reception,40,T) exten = 55512XX,2,Voicemail() So anything coming in with a dialled extension of 55512XX will pattern-match to the above lines. That only works if the caller has dialled one of the other numbers. If we only publish one number (555 1212), what then? -s ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?
the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3 second window to hang up and get his $$ back. This emulates PSTN behavior as close as possible but you have to train your users to hit the DnD button when they walk away from the phonw. -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Monday, June 12, 2006 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line? Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) And zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes callprogress=yes ; define channels context=incoming signalling=fxs_ks channel = 4 Pretty straightforward stuff -- a call comes in on the PSTN line, the Asterisk answers the call, then rings the extension. The caller hears a ring tone throughout the entire process. The rub is that Asterisk has, in reality, taken the PSTN line off hook. Not great if the caller is at a payphone. What if nobody answers the extension? The caller is out his money (50 cents in most of the US, 35 cents in Alberta and 25 cents in the rest of Canada ;) ) So I had the idea of doing things a bit differently, like so: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Dial(SIP/polycom) exten = s,2,Answer() This way, Asterisk dials the extension first, the idea being that when the SIP extension is answered, Asterisk answers the PSTN line and connects the channels. This did not have the expected result -- when I tried this, my SIP extension rang, but answering the extension did not result in Asterisk picking up the PSTN line. There is a way of doing this, isn't there? How can I make this work? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon Europe - Only 1 Week Away
Blogged. Thanks -Dal Asterisk VoIP News - Original Message - From: Steven Sokol [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, June 12, 2006 6:47 AM Subject: [Asterisk-Users] AstriCon Europe - Only 1 Week Away Remember that AstriCon Europe kicks off in only a week with the opening in Berlin. Other events follow in Paris and London. Join us and get to know the Asterisk community in person. We hope to see you there. For more info or to register: http://www.astricon.net Thanks, Steve -- Steven Sokol AstriCon 2006: http://www.astricon.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TTS engine query
Ok Thanks. I just registered 'Diane' also. She seemed to have the best voice. I am curious if you added the Cepstral app or used the festival method described in the Cepstral FAQ. I recompiled with Cepstral app and saw that later. App seems to work fine here. Doug On Mon, 12 Jun 2006, Doug Lytle wrote: Doug Crompton wrote: Not being very happy with festival I would like ro get a better TTS engine. I looked at the listings at: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international but I would like to get user input on suggested packages for Linux. Best performance vs. cost I didn't see Cepstral in that list. www.cepstral.com This is commercial software, I just registered Diane. $29.95 USD. I also see that they charge for a currency license. But, if it works well, we'll probably get a few licenses. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk on AMD 64 BIT
Hey Does asterisk works well on an AMD 64 bit processor server. are there any issues with this ? Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP/SS7 gateway on Sun Ultra 20 amd64
Hello, I have to setup a IP/SS7 gateway on a Sun Ultra 20 Debian Sarge for AMD64 Can we compile asterisk on AMD64 processor ? Harry __ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?
Colin Anderson wrote: the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. That assumes that you are routing to voicemail. That doesn't always apply. Also -- the payphone behaviour varies quite a lot by service provider. I can tell you that in southern Alberta, there is no 2 or 3 second window. When the called line goes off hook, your coins are gone. This is Telus, remember. We're lucky they give us payphones at all. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3 second window to hang up and get his $$ back. This emulates PSTN behavior as close as possible but you have to train your users to hit the DnD button when they walk away from the phonw. Asterisk is so flexible I find it hard to believe there is no way to tell the Zap interface to answer when the corresponding SIP extension is picked up. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] use AT320 international call
Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example 011862, when I actually ment to dial 0118620. Thus left the remaining numbers 0 unsent. The handset had its dial plan disabled. It configured to use iax protocol. My extensions.conf has this: exten=_01186.,1, dial(SIP/voipprovider,60) and it works fine with other iaxy and Cisco ATA. Anyone encounter this symptom? Can you share your experience? Thanks, Min ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DID channels as incoming hunt group?
Colin Anderson wrote: no from the Asterisk perspective it will work regardless of the number dialled as long as it matches the 55512XX pattern. As others have pointed out though, it's just easier to have a single DID and your provider allow multiple channels or instances of the same number to hit your box. As far as I can tell, that's the only thing that will do what I want it to. Even if we have three DIDs, we'd only be publishing one, which would mean that nobody would ever dial the other numbers -- processing calls to those numbers doesn't really help us. But hey, if I can have multiple channels for a single DID, the problem is solved. Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?
[incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer() exten = s,2,Dial(SIP/polycom) Try this exten = s,1,Dial(SIP/polycom,20) exten = s,2,Hangup() I think this way, * won't answer the line until your SIP phone answers. If you don't pickup the phone after 20 seconds it will just ignore this incoming call hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk on AMD 64 BIT
I have 2 servers currently running 64 bit SuSE 10.x on AMD Opteron processors both of which are working very well. John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 On Monday June 12 2006 1:39 pm, Kanishka Somaratne wrote: Hey Does asterisk works well on an AMD 64 bit processor server. are there any issues with this ? Regards Kani -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] get value from DB directly
Search the wiki for the application command realtime() if you are using realtime. www.voip-info.org -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of unplug Sent: Monday, June 12, 2006 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] get value from DB directly Hi, I want to know how I can get a value from a table. Say, I have a table sip_buddies for storing sip user account information. There is a field called 'accountcode' that I want to get its value in the dial plan. As I find that there is no direct way to get the value from the table. Does anyone can tell me how can I get its value in the dial plan? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM Fax Problems
Turn off echo can for those calls. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Monday, June 12, 2006 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM Fax Problems I am running into errors in faxes received by my * server that I route to fax machines on a channel bank. I have a channelized T-1 coming in and I take certain channels and transfer them to channels on my channel bank. The T-1 and channel bank are on a Sangoma A104D 4 port with echo canceling. The faxes tend to have problems with sections of the transmission getting garbled or lost. Does anyone have any information on why this might be happening or how to fix it? -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
