similar to: CallerID fail with Voicetrading operator

On Thursday 18 June 2020 at 19:57:03, Administrator wrote: > does some people here use https://voicetrading.com which is a Dellmont > service from Netherlands. At the high begining they were also known as > Finarea (CH and DE mixed Co) > Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or > equal to CALLERID(num). We tried replacing + with 00, same problem. >

We have two servers but looks like G729 issues. Works fine on the old server and not sure if it is T1 related. See SIP Debug. Any experiences to share. Thanks --- Newark1*CLI> <--- SIP read from 194.xx.Xx.Xx:5060 ---> SIP/2.0 183 Session progress Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=xxxxK784d2637;rport From: "Cell Phone DC" <sip:202xxxxxxx at

Jon, We are only using FastAgi. On the second system (running Asterisk 16) there are no agi's running (just some bash scripts on call hangup). I did add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep -v udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out the packet queue (with the help of

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

VoipRaider the site, says calls to landlines in Brazil is FREE within the freedays period. Log in to the website and hire the service, it says that I have 90 days of freedays paying for cheaper service is $ 10.. That is from what I understand, I will pay 10 dolares for unlimited call in landlines for a period of 90 days? Is that it? Has anyone tested it there? How many simultaneously calls can

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

Hi, sorry for this a bit OT. I'm using VoiceTrading for some calls -premium route- and can't get CID to work despite the fact that CALLERID(num) and CALLERID(name) are setted. I ask in VT->myAccount to accept calls from my IP without checking username & secret. On incoming calls the CID is setted to 0100000000 If I accept calls from username & secret and no IP relation,

Here's a little story on all the cheap guys trying to get the best rate on any route out there ( lcr and others). Anyone have 0.000001 to Mexico billed 1/1 ????? " When customers call us to ask if we sell Cuba termination for 50c/min, I sometimes joke and tell them "sure, I'll give you Cuba for 1c/min, but it will have 0% ASR". Or some other times, I

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of "8|." to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for 'whatever@provider.tld' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hi, I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx. I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I