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 '0049351111111' is now UNREACHABLE! Last qualify: 0

In the CLI I can see:

Name/username Host Dyn Nat ACL Port Status 0049351111111/00493511111 D 5060 UNREACHABLE 0049351222222/00493512222 D 5060 OK (17 ms) 0049351333333 (Unspecified) D…

Asterisk Users 2 months ago 17 Answers

Asterisk As "Proxy" And More Device For A Number


Hi list!

I'm very new in Asterisk and VoIP, and of course I have a problem... :)

Well, my problem is, that Deutsche Telekom wants me to change my ISDN to VoIP... :( I must do that, since I have no alternative.

Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can configure my two numbers by Deutsche Telekom and I got now an extra number from

Now the problems: 1) It seems that I can't configure my ST2022 to have two profiles and both are running on different servers 2) I want that when a number will be called, both…

Asterisk Users 2 months ago 5 Answers

Strange And Complete Failure Of Asterisk 1.8


Hi all

We've had a very strange failure on an Asterisk 1.8 install that has been running for about a year at a customer site.

The physical hardware is fine, all other services off the CentOS 6.5 server are running. Only Asterisk is not working...

The first symptom was that no calls can be made over the SIP phones used with it, and no calls could be received over the SIP trunk connected to it.

I checked and noted that

sip show peers

in the CLI would either do nothing (e. g. just show asterisk*cli> again, with no response) or it would return only this:


Asterisk Users 2 months ago 4 Answers

Too Many Open Files - 786 000 Already Specified As Max Num Open Files?


Hi guys

I have a site on Asterisk running in CentOS 6.5 that has about 150 concurrent callers.

I keep getting these types of messages in the CLI:

[May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n [May 21 11:39:21] WARNING[18469]: chan_sip.c:7041 sip_new: Unable to allocate AST channel structure for SIP channel [May 21 11:39:21] WARNING[18469]: res_rtp_asterisk.c:459 create_new_socket: Unable to allocate RTCP socket: Too many open files [May 21 11:39:21] ERROR[18469]: acl.c:706 ast_ouraddrfor: Cannot create socket

I have specified this on the commandline:

ulimit -n 768000

I have also tried

ulimit -n 512000


Asterisk Users 3 months ago 0 Answers

Monitoring SIP Service (Jai Rangi)


Interesting approach.

What we've done is to write an app that runs on a separate machine that simply does some asterisk -rx calls to the running Asterisk instance via an SSH library and then evaluate the string returned.

For example, to monitor our registered SIP service providers, we compare the output for the sip show registry CLI command when the system is in the correct state, to the output we get on timed checks every five minutes.

If a SIP trunk goes down, the just-obtrained string does not match the stored, correct string anymore, and our app starts sending SMSes / raising alarms.


Asterisk Users 3 months ago 0 Answers

Monitoring SIP Service


Very common concerns from new Asterisk, Freeswitch, opensips and freepbx owners, How can we monitor asterisk, what happens if service stop responding. Here is a small howto on monitoring asterisk with nagios. I am sure there are plenty of options and suggestions, but this is one of them and has been working out very well for us for years.

Best, -Jai

Asterisk Users 3 months ago 0 Answers

Chan_ooh323 To Sip , No Connected Line Info



We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323.

Connection can be shown as


When I do call as avaya user I see name of remote end avay user, i.e. connected line info.

As I see in debug remote side send is as

14:07:29:758 Received H.2250 Message = { 14:07:29:758 protocolDiscriminator = 8 14:07:29:758 callReference = 47 14:07:29:758 from = destination 14:07:29:758 messageType = 7 14:07:29:758 Display IE = { 14:07:29:758 Disa 14:07:29:758 }

over h323.

But now we need to connect to another asterisk over SIP.

in this case we have…

Asterisk Users 3 months ago 1 Answer

Realtime Peers, Mailbox And MWI Problem


Hello, I am facing a problem I can't understand. I have several realtime SIP peers and from time to time, the mailbox field is not loaded in asterisk memory. The mailbox field is correctly populated in the database, but often, after an asterisk restart, the mailbox is not associated to the peer (just to understand, if I run "sip show peer 104-TEST", I see the Mailbox empty. If I run the "sip show subscriptiona", I don't see any subscription for the MWI but only for the BLF.

Is there anyone facing the same problem? How have you solved it?


Asterisk Users 3 months ago 0 Answers

"Retransmission Timeout" Results In Dropped Calls After 32 Seconds



I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this happening in the asterisk console:

This problem only happens a fraction of the time, so I have been unable to enable SIP debugging before it happens to get a capture. However, usually the caller will…

Asterisk Users 3 months ago 15 Answers