similar to: Help with Sip Registration

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

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

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>

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

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:

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:

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

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Hi, just a few days ago I upgraded to openssh-2.3.0p1 on HP-UX 11.00, trusted system, incl. the PAM-patch PHCO_22265 and I run into another problem. The situation is the following. I come from server inside a firewall and go through the firewall wia a ssh-plug-gateway to one host in our internet section and further do a second server in the internet section. All this including the ssh-agent

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 ? --------------

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

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