similar to: SIP 420

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

This is also on Stack Overflow… It’s odd, because sometimes it seems to wrk, and sometimes not??? error.log [2019-01-01 13:08:31] INFO admin/admin_handle_request Bad or missing password on admin command request (command: stats.xml) [2019-01-01 13:08:31] INFO admin/admin_handle_request Bad or missing password on admin command request (command: stats.xml) [2019-01-01 13:08:31] INFO

useRs, Here is some R-ready data for my question to follow. Of course this data is small snippet from a much larger dataset that is about a decade long. Q<-read.table(textConnection("2002 3 28 15 77.38815 2002 3 28 30 77.09505 2002 3 28 45 76.80196 2002 3 28 60 76.50887 2002 3 28 75 76.50887 2002 3 28 90 76.50887 2002 3 28 105

Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. But later it says "Scheduling destruction of sip dialog xxxx ". Then it says "Really destroying sip dialog xxxx". What to do for this problem??? I

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc?.. Typically a reboot of the phone resolves the problem?person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Strange?? Thanks

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

All; In a linked server environment, running Asterisk 1.6 I am noticing that when a call is placed from server A to server B (via 4 digit extension) and server B ext has a FMFM to call their mobile, the mobile phone shows the default caller ID setting on server B instead of the actual caller id of the person who initiated the call on server A. This scenario, of course, works in the event a call

I set up my * server using its publc IP address. Now that i switch over to using the domain name, X-Lite can't log in. =========With Domain Name (doesnt work)============ Transmitting (NAT) to 85.250.206.46:6007 <http://85.250.206.46:6007>: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.250.179.93 <http://85.250.179.93>

I?ve started getting this warning, and nothing I do seems to solve it? INFO admin/admin_handle_request Bad or missing password on admin command request (command: stats.xml) It repeats about every 15 secinds? Any ideas anyone Robert

I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:

Hello I am starting to learn AGI. I have setup an extension to play the agi-test.agi perl script and the output I get is this on console: On Polycom 300: -- Executing Answer("SIP/200-72d2", "") in new stack -- Executing AGI("SIP/200-72d2", "agi-test.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi --

The SIP trunking service that I am trying to set up keeps saying that my registration from Asterisk is invalid. Asterisk registration: REGISTER sip:{registration_ip} SIP/2.0 Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport Max-Forwards: 70 From: <sip:{registration_user}@{registration_ip}>;tag=as5579cc0c To: <sip:{registration_user}@{registration_ip}> Call-ID:

Hi list, How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent from the phones? The trace looks like: ## PHONE -> ASTERISK ## PUBLISH sip:1001 at example.com SIP/2.0 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u;rport From: "1001" <sip:1001 at example.com>;tag=98slbhbn16 To: "1001" <sip:1001 at example.com> Call-ID:

Hi, the following problem occurs on Solaris 2.6. openssh-3.7p1 and openssh-3.7.1p1 both show the same behaviour. openssh is configure with: CC='gcc -L/usr/LOCAL/lib -I/usr/LOCAL/include' ./configure --prefix=/usr/LOCAL --sysconfdir=/etc/ssh --sbindir=/usr/local/sbin --libexecdir=/usr/local/libexec --with-pam --with-tcp-wrappers --with-ssl-dir=/usr/LOCAL/ssl

Hi, I just setup a VoipJet test account (one with 25c credit) to test, they seem to offer good rates to 02 Uk mobiles :) Anyway, everything went ok, iax.conf amended and extensions.conf too, however when I try to make a call I see:- rt*CLI> -- Executing SetCallerID("SIP/2008-d747", "4153574000") in new stack -- Executing Dial("SIP/2008-d747",

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

Now that I have a working kvm setup, I thought I'd finally try how Zimbra works. This is mainly some microbenchmarking, so it may not have much to do with actual performance in real life. Setup: - 1GB memory given to kvm (from host's 2GB) - Intel Core 2 6600 (kvm uses only one CPU) - CentOS 5 - 15GB qcow2 image on XFS filesystem - Zimbra 5.0 RC2 RHEL5 x86_64 - Dovecot latest hg,