similar to: Asterisk not transcoding between installed codecs

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hey, You know, the Digium guys said both are good. They said the the DNS method is better because you dont have the extra point of failure (SER) but said the SER method is better in that it gives you more exact control in the handling of the calls and registration. They did acknowledge there would be a possible downtime only for incoming calls to users with dynamic IPs if the

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry into my

Hi All, In my set-up, I''ve got a cron job that triggers a Puppet run every 20 minutes. I''ve found that on approximately 13 nodes (out of 166), puppetd just hangs. I have to go in, kill the process, remove /var/lib/puppet/state/puppetdlock, and run puppet again and then it''s fine. After a while, it just hangs again so I have to go in, kill the process, etc. Any ideas?

Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

Hi All, I've recently got a "second" number installed on my PSTN line, trusting the Asterisk distinctive ring detection would work as expected. It appeared to work fine at the start, as the second number generated a different ring pattern to 0,0,0 (in the console) only to realise that almost every phone call to this "second" number generated a different ring pattern.

If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing

Hi All, I''ve downloaded a Puppet module and I''m trying to add a parameter to it by editing lib/puppet/type/<resource>.rb. I simply added: newproperty(:pcfree) do desc "My description here" end In the corresponding file in lib/puppet/provider, I do something with :pcfree. Whenever I call the resource from Puppet with my new parameter, it keeps

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

> -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] > Sent: Thursday, March 16, 2006 8:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on > *HAandPolycomphone!! > > > > > > "Q: What are the plans for HA? > > That's BS. Last time I

Hi All, I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree where the users will each have their account, SIP username/password, extension number, context, etc. My first question is: can this be done with 1.4.x? If so, where can I get the res_config_ldap from?? I googled quite a bit and found a res_config_ldap that looks to be coded for 1.2. Is anyone running

Hi guys, I'm interconnecting an Asterisk box with a Lucent Definity PBX by means of FXO/FXS ports on a TDM2400 card. Everything works well, except for one little thing. Every now and then somebody (from an Asterisk extension) will call another extension on the Lucent Definity PBX and they hit their voicemail. They caller leaves their message (or not) and hangup, BUT the Lucent sometimes