Re: [asterisk-users] Best Scripting Language
Am 01.04.2011 14:27, schrieb Roger Burton West: On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Depends on the other parameters. Perl is great for rapid development, but I wouldn't run it per-call on a box taking hundreds of calls per second. (Ditto Ruby and Python.) C will be much faster, but it's more effort to write and debug. Another solution could be a combination of PHP and HipHop. Easy to develop and after transaltion with HipHop very performant. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hold problem with Queue
Sorry, during the weekend I don't have access to logs.
srvcom*CLI core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
24 calls processed
=== SIP/1CDF0F4AD346 call 3600
== Using SIP RTP CoS mark 5
-- Executing [3600@interne:1] Set(SIP/1CDF0F4AD346-0041,
CALLERID(name)=APPEL DE TEST) in new stack
-- Executing [3600@interne:2] Set(SIP/1CDF0F4AD346-0041,
CALLERID(num)=NE PAS REPONDRE) in new stack
-- Executing [3600@interne:3] Answer(SIP/1CDF0F4AD346-0041,
) in new stack
-- Executing [3600@interne:4] Queue(SIP/1CDF0F4AD346-0041,
TestQueue180) in new stack
-- Started music on hold, class 'default', on SIP/1CDF0F4AD346-0041
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- SIP/1CDF0F4A35F2-0043 is ringing
[Apr 4 09:01:10] WARNING[8440]: mp3/interface.c:216 decodeMP3: Junk
at the beginning of frame 49443303
-- SIP/E05FB9818972-0044 is ringing
-- SIP/002699ABE031-0042 is ringing
-- Nobody picked up in 1 ms
-- Nobody picked up in 1 ms
-- Nobody picked up in 1 ms
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- SIP/1CDF0F4A35F2-0046 is ringing
-- SIP/E05FB9818972-0047 is ringing
-- SIP/002699ABE031-0045 is ringing
-- SIP/002699ABE031-0045 connected line has changed. Saving it
until answer for SIP/1CDF0F4AD346-0041
-- SIP/002699ABE031-0045 answered SIP/1CDF0F4AD346-0041
-- Stopped music on hold on SIP/1CDF0F4AD346-0041
-- Locally bridging SIP/1CDF0F4AD346-0041 and SIP/002699ABE031-0045
srvcom*CLI core show channels
Channel Location State Application(Data)
SIP/002699ABE031-000 3600@interne:1 Up AppQueue((Outgoing Line))
SIP/1CDF0F4AD346-000 3600@interne:4 Up Queue(TestQueue180)
2 active channels
1 active call
25 calls processed
=== SIP/002699ABE031 Hold SIP/1CDF0F4AD346
=== SIP/002699ABE031 Hangup
srvcom*CLI core show channels
Channel Location State Application(Data)
SIP/002699ABE031-000 3600@interne:1 Up AppQueue((Outgoing Line))
SIP/1CDF0F4AD346-000 3600@interne:4 Up Queue(TestQueue180)
=== If SIP/1CDF0F4AD346 want hold SIP/002699ABE031, it's work
-- Started music on hold, class 'default', on SIP/002699ABE031-0045
Thanks for your help
2011/4/1 Satish Patel satish...@hotmail.com:
We need logs or console output
--
Sent from my iPhone
On Apr 1, 2011, at 9:01 AM, Elensarde elensa...@gmail.com wrote:
Yes, when the caller are in the queue
New informations :
- If A call B directly and B hold A, it's work...
- Test with Asterisk 1.8.0, 1.8.1, 1.8.2, same problems...
- Phones : Cisco SPA502G / SPA508G / SPA509G
2011/4/1 Satish Patel satish...@hotmail.com:
Do you have music on hold configure?
--
Sent from my iPhone
On Apr 1, 2011, at 3:39 AM, Elensarde elensa...@gmail.com wrote:
Hello List,
First, sorry for my bad English skill, I'm French.
We have an asterisk 1.8.3.2 built from sources with a simple Queue :
[TestQueue]
strategy=ringall
timeout=15
retry=1
timeoutpriority=conf
ringinuse=yes
wrapuptime=2
member = SIP/002E31,0,Agent A
member = SIP/1CA3F2,0,Agent B
member = SIP/E08972,0,Agent C
And this dialplan (extension.ael) :
3600 = {
Answer();
Queue(TestQueue60);
Playback(invalid);
Hangup();
}
When somebody call this exten, an Agent take the call without problems.
But when he want hold this, phone try to hold the caller without
success.
Finally, no signal in the caller-line and the agent-line is hangup
(for the phone), I not have errors or warnings in logs...
Any ideas ?
Thanks in advance, and kind regards,
Elensarde
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[asterisk-users] SIP channel able to add codecs once up and running?
From my observations, if a video capable device starts the call in non-video mode, it is never able to add video to the channel? Is this correct, or am I missing something? It looks as if the codec 'jointcapability' is calculated at the start of the call, and can never be added to (with exceptions for T.38 fax) as any SDP update is masked using the existing 'jointcapability' and knocks out the newly requested codec. Is that right? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Finding out asterisk settings from console
Hi In asterisk 1.4, is there a way I can find out what default settings the server is using that would normally be changed in the asterisk.conf? I've had a look through the command list and nothing seems to do what I'm after, was really hoping to see something like core show settings like you get with sip show settings but there isn't one. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI detection
NT = Network Termination/Topology (or something like that) - used when you want to be the network end. TE = Terminating Equipemt - used when you want to be the consumer end (a PBX or ISDN handset usually). You probably want to be the TE - as you are running Asterisk PBX ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta Sent: 01 April 2011 13:37 To: asterisk-users@lists.digium.com Subject: [asterisk-users] BRI detection Hi, I need to configure BRI 4span card in dubai in vicidialnow for dialer perpose. in that i have small confusion which is NT an TE mode . that was i am setting perfectly but dubai telco what they are use for this i dont know which parameters are use for that . please help me. -- Best Regards, Mahesh Katta BUZZWORKS Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 303, Gagangiri Apts, Parleshwar Road, Ville Parle East, Mumbai - 400057. GSM +91.97029.70779 | Phone +91.22.2663.1811 | Fax +91.22.2663.1811 Web http://www.buzzworks.com If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR fields not being written from h extension after Dial command completes.
Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 Arjan Kroon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crashes on high IO load
Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load Asterisk Module with parameters?
Hi List I´m not sure if this the right list for my question but i hope you can help me. I have write on a Asterisk Module and have the *.so File. Load this Module once is no problem , but I want to start it with a parameter multiple times. so it looks like this module load modul.so connect=programm_a module load modul.so connect=programm_b and then it must be run in two instances. Is it possible or must i compile it into two different modules and load then in 2 steps? module load modul.so module load modul2.so Best Regards Meckel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.
On Mon, 2011-04-04 at 13:58 +0200, Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 Arjan Kroon Hi Have you set endbeforehexten=yes in your cdr.conf? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Asterisk Module with parameters?
On 04/04/2011 08:36 AM, Meckel wrote: Hi List I´m not sure if this the right list for my question but i hope you can help me. I have write on a Asterisk Module and have the *.so File. Load this Module once is no problem , but I want to start it with a parameter multiple times. so it looks like this module load modul.so connect=programm_a module load modul.so connect=programm_b and then it must be run in two instances. Is it possible or must i compile it into two different modules and load then in 2 steps? module load modul.so module load modul2.so There is no facility in Asterisk for parameters to be passed to a module. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.
Hi, I tried both setting (yes and no), both with the same result. Greeting, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Ishfaq Malik Verzonden: 04-04-2011 15:53 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. On Mon, 2011-04-04 at 13:58 +0200, Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 Arjan Kroon Hi Have you set endbeforehexten=yes in your cdr.conf? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call forwarding
Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XX the call will be forwarding automatically to anther number 0520xx Does anybody have a solution to this problem. Thanks and Regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashes on high IO load
Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan matching
Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) exten = _01137455.,2,Goto(intl-disabled,s,1) exten = _01137477.,3,Goto(intl-disabled,s,1) exten = _0113749.,4,Goto(intl-disabled,s,1) exten = _011.,5,Goto(intl-disabled,s,1) exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Asterisk Module with parameters?
On 04.04.2011 15:55, Kevin P. Fleming wrote: On 04/04/2011 08:36 AM, Meckel wrote: Hi List I´m not sure if this the right list for my question but i hope you can help me. I have write on a Asterisk Module and have the *.so File. Load this Module once is no problem , but I want to start it with a parameter multiple times. so it looks like this module load modul.so connect=programm_a module load modul.so connect=programm_b and then it must be run in two instances. Is it possible or must i compile it into two different modules and load then in 2 steps? module load modul.so module load modul2.so There is no facility in Asterisk for parameters to be passed to a module. Thx for this information -- ___ Ulrich Meckel Entwicklung Mail: mec...@netzing.de NETZING Solutions AG Tel.: 0351/41381 - 0 Fröbelstr. 57, 01159 Dresden Fax: 0351/41381 - 12 ___ Impressum: NETZING Solutions AG - Fröbelstraße 57 - 01159 Dresden Sitz der Gesellschaft Amtsgericht Dresden HRB 18926 Vorstand Dieter Schneider - Aufsichtsratsvorsitzender Volker Kanitz USt.Id DE211326547 Mail: netzing...@netzing.de -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to check if the call is using t38 except in the sip packets
How could i check if the call is using t38 except looking at the sip debug? Is there any variable thats set or could be set? thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan matching
On Mon, Apr 4, 2011 at 8:09 AM, Asterisk User asteruserl...@gmail.comwrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) exten = _01137455.,2,Goto(intl-disabled,s,1) exten = _01137477.,3,Goto(intl-disabled,s,1) exten = _0113749.,4,Goto(intl-disabled,s,1) exten = _011.,5,Goto(intl-disabled,s,1) exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. Assuming that 011870 is followed by more than digit, normally, I'd say your first set is more applicable. The . in the pattern at the end means any number of digits, followed by a timeout. If you know the number of digits, and it is fixed, then you could use _011870XXX or similar to avoid the timeout, and begin the Goto immediately on reception of the final digit. The X in the second set will match just one digit, and the Goto will be be executed. Does that help? -- Steve Murphy ParseTree Corporation -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP register and contact header
Hello, I define SIP registrations as follow in sip.conf : register = number:passwd@sip-server example : register = 33:mypass@ip_sip_server But apparently the SIP 'contact' header in the SIP REGISTER looks like this : /Contact: sip:s@ip_my_asterisk/ How come ? And how to change this so it reads : /Contact: sip:/33/@ip_my_asterisk/ Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashes on high IO load
Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?
No problem. You just specify accountn...@gmail.com. exten = accountn...@gmail.com,1,Answer() exten = accountn...@gmail.com,n,Wait(2) exten = accountn...@gmail.com,n,SendDTMF(1) exten = accountn...@gmail.com,n,Dial(SIP/devicename) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Friday, April 01, 2011 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers? Hello. I would like to configure Asterisk to accept incoming calls from two different GoogleVoice numbers via gtalk and jabber. I'm running Asterisk 1.8.3.2 and I can get one number working just fine. However, I can't figure out how to modify the gtalk.conf file shown on the Asterisk wiki site to work with two different jabber profiles. Do all incoming GoogleVoice calls have to go through the [guest] context in gtalk.conf? If so, it seems that would limit you to working with only one GoogleVoice number. My configs basically match what's at the wiki site here: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google Any advice would be appreciated. Thanks! -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashes on high IO load
Are you using IAX? There are some problems causing crashes for us related to laggyness on IAX channels with 1.8 versions. There are a bunch of problems with IAX related to https://issues.asterisk.org/view.php?id=17521 Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Maximilian Grobecker Sent: 04 April 2011 16:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk crashes on high IO load Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?
On Mon, Apr 4, 2011 at 10:25 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: No problem. You just specify accountn...@gmail.com. exten = accountn...@gmail.com,1,Answer() exten = accountn...@gmail.com,n,Wait(2) exten = accountn...@gmail.com,n,SendDTMF(1) exten = accountn...@gmail.com,n,Dial(SIP/devicename) Thanks for the reply. I did get it working over the weekend by modifying my jabber.conf to include both of my gmail accounts. Asterisk can now receive incoming calls from both of my googlevoice numbers. My outgoing calls use the context that I specified in gtalk.conf. It works just fine for me. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding
Do this:
exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)
you can also use the dial command for this as well
exten= _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})
replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
contains 0520 numbers.
I have not tested it, you can try it on your setup.
On Mon, Apr 4, 2011 at 7:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list,
i have one question related to call forwarding.
i have 2 number for the inbound and i want to configure asterisk like that.
When the customer call the first number 0522XX the call will be
forwarding automatically to anther number 0520xx
Does anybody have a solution to this problem.
Thanks and Regards.
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E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Dialplan matching
On Monday 04 Apr 2011, Asterisk User wrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. Asterisk's default behaviour is always to try the hardest-to-match expression first (i.e. the exact extension number). If there is no match there, it then tries progressively easier-to-match expressions; only ever trying something like _. if nothing else matched. (Compare the rules of poker when wild cards are introduced: a natural hand beats an otherwise-equivalent hand containing wild cards.) -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call-limit bypass
Hi everyone, one of our users last night bypassed asterisk call-limit limitation. I have no Idea how. Is it possible? Is there a bug in asterisk that can be manipulated for this purpose? The call-limit variable was to 2, and the user initiated 169 calls in 2 minutes each has duration at least 8 minutes. Please comment... Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI-Linux 2.4.1.1 Released
The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.1. DAHDI-Linux 2.4.1.1 and DAHDI-Linux-Complete 2.4.1.1+2.4.1 are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete 2.4.1.1 is a maintenance release to fix a regression in DAHDI-Linux 2.4.1 from DAHDI-Linux 2.4.0 which prevented the LEDS on the TDM410 and AEX410 panel from lighting when the wctdm24xxp module loads (#18939) [1]. [1] https://issues.asterisk.org/view.php?id=18939 Issues found in these releases can be reported in the DAHDI-linux project at https://issues.asterisk.org Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe headache
Ok, I've been running applications on 1.4 for quite some time using meetme to hold a person, while the person on the other end of the call accepts, etc. I was playing status messages to the calling party using a context like this: [status-one-en] exten = 100,1,Playback(my_status_message) exten = 100,1,Hangup() and then creating a call file like this: Channel: Local/100@status-one-en CallerID: Rick 55 MaxRetries: 0 RetryTime: 15 WaitTime: 45 Application: MeetMe Data: 12345,qdM and it would hook into the meetme, play the message, then hangup and drop out. I've been building an application with 1.6, and this isn't working at all. In verbose mode, I see the message played, and the call hang up, but the music never even stops on the meetme. After about 20 seconds I get: Call failed to go through, reason (3) Remote end Ringing Is there some other way to do this in 1.6 that I'm unaware of? I've tried creating a context and extension for the meetme portion (rather than using the Application/Data in the call file, and switched the order around (which does cause the music to stop, but the announcement still doesn't get played, and I get the same call failed message). I've been googling on this for days now, and really just need to get it working. TIA Rick CONFIDENTIALITY / PRIVILEGE NOTICE: This transmission and any attachments are intended solely for the addressee. This transmission is covered by the Electronic Communications Privacy Act, 18 U.S.C §§ 2510-2521. The information contained in this transmission is confidential in nature and protected from further use or disclosure under U.S. Pub. L. 106-102, 113 U.S. Stat. 1338 (1999), and may be subject to attorney-client or other legal privilege. Your use or disclosure of this information for any purpose other than that intended by its transmittal is strictly prohibited, and may subject you to fines and/or penalties under federal and state law. If you are not the intended recipient of this transmission, please DESTROY ALL COPIES RECEIVED and confirm destruction to the sender via return transmittal. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] From 1.4 to 1.8: stdexten issue
On Sun, Apr 03, 2011 at 10:35:52PM +0200, Benny Amorsen wrote: stdexten in the default extensions.conf seems to only handle extensions with at least 2 digits... Good one, I hadn't noticed that. Thanks that fixed it!!! -- Mathieu Chouquet-Stringer math...@csetco.com The sun itself sees not till heaven clears. -- William Shakespeare -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan matching
On Monday 04 April 2011 09:09:28 Asterisk User wrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) This one is okay. exten = _01137455.,2,Goto(intl-disabled,s,1) Change this to priority 1. exten = _01137477.,3,Goto(intl-disabled,s,1) Change this to priority 1. exten = _0113749.,4,Goto(intl-disabled,s,1) Change this to priority 1. exten = _011.,5,Goto(intl-disabled,s,1) Change this to priority 1. exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup This looks like it should be starting from priority 1, extension s, context [intl-disabled]. Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. The major problem in your dialplan is that you WANT to have multiple start points, but the way you have it written, there is only ONE start point. Everything else is simply ignored. Extensions will only start in the dialplan from priority 1. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi
very thanks, that's work
bye
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawskimarkm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes
the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten}
])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten =
_00339,7,Set(CALLERPRES()=prohib_not_screened)
exten =
_00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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Re: [asterisk-users] call-limit bypass
From what I understand on the newer versions of asterisk call-limit does not limit calls anymore. You have to limit them from your code using call groups. From what I have seen on the 1.6x and 1.8 versions call-limit does not limit your call counts. We use code and the GROUP_COUNT to limit calls. If you use it right it is rock solid. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Rizwan Hisham rizwanhas...@gmail.com Sent: Monday, April 04, 2011 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] call-limit bypass Hi everyone, one of our users last night bypassed asterisk call-limit limitation. I have no Idea how. Is it possible? Is there a bug in asterisk that can be manipulated for this purpose? The call-limit variable was to 2, and the user initiated 169 calls in 2 minutes each has duration at least 8 minutes. Please comment... Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan is not finding my number asterisk 1.8.3
I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the dialplan show it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] Its telling me it cannot find it. Its there - the dialplan shows its there. When I stop and start it works again for a little while. Matter of fact I just issued dialplan reload and calling into 1105 works again. Whats up? How do I get this to be consistent? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Hi Satish! Few days ago I had the same problem, and was a problem in my dialplan. Post your extensions.conf and let's see. Best regards, Fellipe From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 19:51:26 + Subject: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Satish, Run dmesg and look for anything funny. This sounds very similar to when I had a netfilter nat helper not helping me at all. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Satish, Run dmesg and look for anything funny. This sounds very similar to when I had a netfilter nat helper not helping me at all. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP register and contact header
On 4/4/11 5:13 PM, Jonas Kellens wrote: I define SIP registrations as follow in sip.conf : register = number:passwd@sip-server example : register = 33:mypass@ip_sip_server But apparently the SIP 'contact' header in the SIP REGISTER looks like this : /Contact: sip:s@ip_my_asterisk/ Change your register line into this: register = 33:mypass@ip_sip_server/33 -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Read VoiceMail direct
Hey Guy!
I want direct access of VoiceMail without asking mailbox number (Direct ask
PIN). I am using following dialplan but its still asking me Mailbox number.
Look like asterisk 1.8 doesn't support CALLERIDNUM variable.
Do you have any idea ?
exten = 8500,1,answer
exten = 8500,2,wait(1)
exten = 8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
exten = 8500,4,hangup
exten = i,1,playback(invalid)
exten = i,2,hangup
--
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Re: [asterisk-users] Read VoiceMail direct
Hi,
maybe:
exten = 8500,3,VoiceMailMain(${CALLERID(num)}@default)
Regards
- Andrea
- Original Message -
From: satish patel
To: asterisk-users
Sent: Monday, April 04, 2011 11:08 PM
Subject: [asterisk-users] Read VoiceMail direct
Hey Guy!
I want direct access of VoiceMail without asking mailbox number (Direct ask
PIN). I am using following dialplan but its still asking me Mailbox number.
Look like asterisk 1.8 doesn't support CALLERIDNUM variable.
Do you have any idea ?
exten = 8500,1,answer
exten = 8500,2,wait(1)
exten = 8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
exten = 8500,4,hangup
exten = i,1,playback(invalid)
exten = i,2,hangup
--
--
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Re: [asterisk-users] Read VoiceMail direct
Perfect! Thanks
what about :-4 ? I want to remove some digits
-satish
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 4 Apr 2011 23:16:30 +0200
Subject: Re: [asterisk-users] Read VoiceMail direct
Hi,
maybe:
exten =
8500,3,VoiceMailMain(${CALLERID(num)}@default)
Regards
- Andrea
- Original Message -
From:
satish
patel
To: asterisk-users
Sent: Monday, April 04, 2011 11:08
PM
Subject: [asterisk-users] Read VoiceMail
direct
Hey Guy!
I want direct access of VoiceMail without
asking mailbox number (Direct ask PIN). I am using following dialplan but its
still asking me Mailbox number. Look like asterisk 1.8 doesn't support
CALLERIDNUM variable.
Do you have any idea ?
exten =
8500,1,answer
exten = 8500,2,wait(1)
exten =
8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
exten =
8500,4,hangup
exten = i,1,playback(invalid)
exten =
i,2,hangup
--
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Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.
On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
On Mon, Apr 4, 2011 at 2:20 PM, Jerry Geis ge...@pagestation.com wrote: snip Whats up? How do I get this to be consistent? Jerry Can you post all of the relevant sections of extensions.conf, and the CLI output of a successful call and the CLI output of a failed called. The complete CLI output, from beginning to end of each call. With this kind of information we can begin to diagnose what's happening. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
On 11-04-04 03:20 PM, Jerry Geis wrote: I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the dialplan show it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] *CLI dialplan show 1105@smvoice-mediaport Its telling me it cannot find it. Its there - the dialplan shows its there. When I stop and start it works again for a little while. Matter of fact I just issued dialplan reload and calling into 1105 works again. Whats up? How do I get this to be consistent? Have you included the context properly? [mndemo_to_mediaport105] include = smvoice-mediaport -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Number Conversion
Hi all,
Please, could somebody point me out what is going wrong in this line below?
exten = _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT)
As I know, such line must convert any number dialed to 021, therefore, as we
can see, it's kept the number dialed!
-- Executing [00151236445600@a2billing:1] Dial(SIP/2000-,
DAHDI/G0/0151236445600,45,rT}) in new stack-- Called G0/0151236445600
Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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Re: [asterisk-users] Number Conversion
On 5/04/11 1:00 PM, Flavio Miranda wrote:
Hi all,
Please, could somebody point me out what is going wrong in this line below?
exten = _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT)
As I know, such line must convert any number dialed to 021, therefore,
as we can see, it's kept the number dialed!
It must not be running that line - have you done a dialplan reload?
--
Cheers,
Matt Riddell
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http://www.venturevoip.com/cc.php (Call Centre Solutions)
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[asterisk-users] Number Conversion
I did
That's weird, doesn't it!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Tue, 5 Apr 2011 13:02:06 +1200
From: li...@venturevoip.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Number Conversion
On 5/04/11 1:00 PM, Flavio Miranda wrote:
Hi all,
Please, could somebody point me out what is going wrong in this line below?
exten = _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT)
As I know, such line must convert any number dialed to 021, therefore,
as we can see, it's kept the number dialed!
It must not be running that line - have you done a dialplan reload?
--
Cheers,
Matt Riddell
___
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http://www.venturevoip.com/exchange.php (Full ITSP Solution)
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Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
On Mon, Apr 4, 2011 at 3:20 PM, Jerry Geis ge...@pagestation.com wrote: I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the dialplan show it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] Its telling me it cannot find it. Its there - the dialplan shows its there. When I stop and start it works again for a little while. Matter of fact I just issued dialplan reload and calling into 1105 works again. Whats up? How do I get this to be consistent? Jerry I'm not all that familiar with 1.8 yet but, with 1.6.2, I ran into some similar problems with extenpatternmatchnew=yes. They were similar in that the dialplan was not executed as expected, but the behavior was deterministic. Your experience has things changing over time which is really quite strange. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan is not finding my number asterisk1.8.3
I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com -- Sent from my BlackBerry® Senior Support Engineer US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with Digum Repos / AsteriskNOW Bad Packages
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming kpflem...@digium.com wrote: On 03/23/2011 10:53 PM, Andrew Joakimsen wrote: I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? Yes... but this indicates that the module that was built appears to be broken. I'll let the package maintainer know. Bug 0018748 is closed a few days ago, but I don't see any new RPM... -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check if the call is using t38 except in the sip packets
On 4/04/2011 10:27 PM, Israel Gottlieb wrote: How could i check if the call is using t38 except looking at the sip debug? My suggestion is to capture the packets between the two end points using tcpdump, in your case you would be captuing SIP RTP packtes between your SPA8000 your ITSP. I would suggest you ensure you use a snaplen in tcpdump that is large enough to capture all the packet information. Once you have the captured data you will find Wireshark is your friend, open the file in Wireshark and then select VoIP Calls which is located under the Telephony menu item. Once it has gathered the information it wants you will be able to select one ore more sessions, once selected click on the Flow button, all will now be clear! Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
