similar to: Asterisk crash

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

Hi List, I have faced a problem in asterisk queue implementation. I configured a queue with 'ringall' strategy and 'ringinuse=yes' in queues.conf. If three calls come to this queue in parallel, the logged in queue agent used to get only one call (may be the first one), not all the calls waiting in the queue at a time. Once the agent answers the call the next call is displayed. I

Hello List, Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode enabled as INFO for SIP channel. If it is a bug in asterisk or something need to be enabled in sip.conf for the same. Dialplan looks like Exten => 100,1,MeetMe(100,dmF) Sip.conf dtmfmode=info Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

Hello List, Is it possible to run an AGI script in backgroung for all the associated SIP channels in ConfBridge Application? If yes how? This can be done using 'b' parameter in MeetMe for non SIP channels. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.

Hello List, We are running two asterisk machines in virtual IP as primary and secondary server. Initially virtual IP will be active in primary server; during the failure of primary secondary will get the virtual IP. Is there any way to retrieve pending queue calls from primary to secondary, in case primary fails? Does asterisk provide any interface to do it or we have to write some application

Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran

Hello List, What version of DAHDi should be installed for CentOS Kernel version 2.16.18-194.el5. We do not have access to yum in our network, so we need to install a specific version with respect to kernel version. Or, what update to be downloaded and applied to CentOS kernel to install a specific version of DAHDi. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor,

Hello List, Asterisk CLI command "core show hints" gives the list of hint extension configured and its presence status. In command output there is a field called "watchers" and it contains a numeric value of number of subscriptions' registered for that particular extension. So, is there any CLI command to check who the watchers for an extension are? Regards, Rajib Rajib

Dear Forum, I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing "meetme kick all" CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which release of asterisk this issue has been fixed. Please help. https://issues.asterisk.org/jira/browse/ASTERISK-15741 With best regards, Rajib

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

I am new to R. I have a data like: x y z w p .......... m 1 10 15 20 25 30 2 11 16 21 26 31 3 12 17 18 19 20 4 51 52 53 55 67 ....... thus I have 145 rows

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thank you Paul. I have downloaded the code. How out-of-call messaging can be configured in the Dialplan? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Thu, 07 Apr 2011 10:14:37 -0400 From: Paul Belanger

dear railers I was attempting to display a svg chart inside a tooltip in rails using Scruffy. my browser is Firefox 2 and i am using WEBRICK. when i render inside the controller using graph.render(:size=> [255,205], :to => ''C:\xyz.svg) and serve iit via rhtml using the <embed> tag. Instead of the chart inside the tooltip i get a dialog box asking me to open it ... with

Hello List, I have a requirement like, Only one UA can register at a time (the registration should be independent of IP). If some other UA tries to register from a different IP using the same credentials, it should be blocked by asterisk. We do not want to permit or block any IP or subnet in sip.conf. Following is an example of sip user configuration, [217] type=friend username=217 host=dynamic

Hello List, Please help with the following problem, I have a situation, where I need to play an audio announcement to the caller SIP channel once an attended transfer is successful. The attended transfer is done from client. I can see a transfer event in AMI. I am not using 'T/t' option in dial() command. The transfer is completely on client side using SIP signaling. 1. A calls B 2. B