similar to: Grandstreams randomly go busy with Asterisk?

Dear All I have a problem with a TDM400 card with 4 x FXS modules. The card carries extensions only and there are no incoming lines. I can make a call to the extension on this card with no problems. However, when I try and call out I just get a busy signal. I also get an error message (as shown at the bottom). Is this a problem? Configs below: [root at asterisk etc]# more zaptel.conf

Hi, My local server is running CentOS 7. The machine has two NICs and is acting as a gateway. For DHCP and DNS, I'm using Dnsmasq. I have a strange little problem with local hostname resolution. Before going into more details, here's my configuration. [root at nestor:~] # ifconfig enp2s0: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500 inet 192.168.0.2 netmask

Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c:

How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/Username Host Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server (null) (D) 255.255.255.255 0 Unmonitored voip

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

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

Hello, I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know how can i monitor the extension status? when i wrote sip show peers on asterisk Extension Domain port Status 111/111 (Unspecified) D 0 Unmonitored 1300/1300 192.168.50.111 D 5060 Unmonitored 222/222

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

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

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.

I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 <firewall-ip> D N 255.255.255.255 60665 Unmonitored tp2/tp2 <firewall-ip> D N

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake -------------- next part -------------- An HTML attachment was

When doing a "sip show peers" I might see something like: Name/username Host Dyn Nat ACL Port Status devcentos5x64_to_mmfirepa 192.168.1.177 5060 Unmonitored devcentos5x64_to_bt610tMM 192.168.1.159 5060 Unmonitored devcentos5x64_to_am2mm/de 192.168.1.178 5060

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER