[Asterisk-Users] Asterisk Newbie

Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is noa forum-like tool to search thru the posts by keyworks for example.

[Asterisk-Users] hardware question

Hello I want to to know if the motherboards VIA are fully supporte by asterisk. And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? thank you Fabian

[asterisk-users] T.38 passthru on 1.8.5

Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax

[asterisk-users] same sip peer as user and provider

Hello Up to version 1.6.0 we have been able to configure the same SIP device as a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we switched to version 1.8 this setup wont work, apparently one can not have the same IP on 2 different trunks anymore. The trunk that is

Re: [asterisk-users] same sip peer as user and provider

yes, same thing From: fbo...@hotmail.com To: fbo...@hotmail.com Subject: RE: same sip peer as user and provider Date: Tue, 30 Aug 2011 10:35:01 -0400 yes my friend. same thing From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: same sip peer as user and

Re: [asterisk-users] T.38 passthru on 1.8.5

both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a

Re: [asterisk-users] T.38 passthru on 1.8.5

will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect

Re: [asterisk-users] T.38 passthru on 1.8.5

txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? From:

Re: [asterisk-users] T.38 passthru on 1.8.5

Txs a lot Kevin. I had just created and account on https://issues.asterisk.org/jira Let me know if this is the right place to post both the pcap capture and the sip logs. If not please help me out creating the account in the right place so that I can provide all the information you need. The sip

Re: [asterisk-users] T.38 passthru on 1.8.5

Hi Kevin, I created the issue on the https://issues.asterisk.org/jira web site, posted the description of the prob and submitted asterisk console logs [sip and udptl debug on] and a wireshark capture taken on the asterisk machine showing both legs with signaling and media. PLease let me know

[asterisk-users] Asterisk removes SDP from 180 with SDP

Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting

[asterisk-users] Reply to INVITE with 1 codec

In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to yes the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile. But in version 13.1 (I think version 11.2 also) is not working like that , it keeps sending all the codecs and

[asterisk-users] Question about Warning message

Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function sip_write inside chan_sip.c. In our previous

Re: [asterisk-users] Question about Warning message

thank you, we are using the same configuration files in 13, same setup, just different asterisk version. we just dont see the msgs in the console/logs, it is the same exact voice traffic on both asterisk versions is that something that you set on/off? if that is the case how can it be done?

[asterisk-users] set codec based on B side

Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints: [Alice] disallow:all allow:ulaw,alaw,g729 [Bob] disallow:all allow:ulaw,alaw,g729 Alice calls into Asterisk on ext 100 and then we dial Bob I want to