similar to: Can't connect to SIP provider

Displaying 20 results from an estimated 8000 matches similar to: "Can't connect to SIP provider"

2007 Jul 17
0
help with sip configuration for sipgate.de on asterisk 1.4
hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help
2005 Mar 04
2
budgetphone
Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here
2005 Jul 10
0
(no subject)
I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I'm new to Asterisk I can't get the error why this is not working. To me it all
2005 Jul 10
3
Incoming calls from BudgetPhone.nl
(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not
2005 Jun 05
1
Unable to create channel of type SIP-please help
Hi there, I'm having a hard time getting outbound calling to my SIP-->PSTN gateway. I continuasly get the following result in my log files: Jun 5 10:07:50 WARNING[1568]: No such host: t2y Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP' Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time I make the following context in my
2015 May 29
0
Calling from "extern"
Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in
2005 May 10
0
outbound PSTN numbers over SIP failing
Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in
2005 May 17
4
multiple sip accounts from same sip registrar
Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an
2008 Nov 27
0
trunk peer not registering after migrating installation
I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. "sip show peers" shows my phones
2010 Apr 16
2
Testing a sip call through Asterisk?
I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status:
2016 Mar 15
2
Fwd: Unable to place outbound calls
Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")
2006 Dec 12
0
Disregistering Constantly - message: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0
Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer
2004 Feb 08
1
Registering SJPhone with Asterisk
2008 Jan 11
5
Congestion/Forbidden issue with new carrier
Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new
2013 Apr 06
1
sip registration
I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command "sip show registry" and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the
2006 May 24
0
Dual Line SIP config to the same provider
Hi, I have a setup where I have multiple lines to the same provider - in this case broadvoice. I have created the entries in sip.conf for both accounts - and independently they work fine. When they both are in use, incomming calls are placed to the one that's the last in the sip.conf file. On voip-info I found the following: *Quote:* When Asterisk receives an incoming SIP call, the SIP
2006 Feb 06
1
Will not authenticate incoming VOIP provider calls
I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I
2004 Apr 07
2
error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)