* You are viewing Posts Tagged ‘sip’

Alembic – Asterisk 11

I’ve had years of experience using ODBC for CDR, SIP, and extensions with Asterisk. One thing that has been problematic in the past is with documentation as far as database tables changing between versions (even within minor releases, though that was back in the 1.4 days). I was excited to see there is a plan for better managing that on Asterisk 12 via Alembic. All that being said, are there any plans to implement that with Asterisk 11, since that is the current LTS release? Or are we pretty sure the table structure won’t be changing within that version through the rest of its lifespan, making such an effort a waste?

Thanks,

Josh

No Voice When The Calls Come From Internet

Hi,


I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection.

When I call from the SIP device at home the SIP account on the Internet
(iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I answer I don’t hear any thing!

I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a defined UDP range in rtp.conf forwarded to my Asterisk server.

Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ?

Nevertheless, when both SIP devices are in the same home IP network the call is made without any problem whatever who starts the call.


Regards,

Dialplan To Reach External SIP Phone

If I have Asterisk setup with local SIP phones configured but need to call a SIP phone which is not local but actually on another VLAN, what would the dialplan need to look like?

Handset Forwarding Diversion Header Cannot Be Set On Local Channels

is there anyway to change Sip headers in local channels?
if a user sets forward on their handset, calls coming in to the handset get diversion header added:
Diversion: “202″ ;reason

SIPAddHeader Back To Source

Hi,

I am using the XML-browser and Call-Info header features for some SIP phones. SIPAddHeader(Call-Info: …) seems to work only in the outgoing direction. Does somebody know a way to send a Call-Info header to the originating SIP device by using only the dial plan?
Currently, I am using the XML-browsers to update callee info, but I’d like to use the icon purpose to do that.

It’s probably easier to embed this functionality into a CTI application using an AMI command like Originate (such that internally Dial() gets called twice), but this triggers it from the outside. sipsak could be called from extensions.conf, but I’d like to avoid that. Transferring to a Local channel after entering the dial plan might also work, but that looks clumsy. I am sorry if I have overlooked a standard method to send a header back to the source.

jg

SipML5, Ast12 And WebRTC: Not Acceptable Here

Hi All.

I’m running some tests with the latest Asterisk SVN-branch-12-r410493M
compiled with fresh github pjsip and srtp 1.4.2 on an i386 CentOS
machine (2.6.32-358.18.1.el6.i686). As a client I’m using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m.

I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call from the softphone, Asterisk answers with a 488 not acceptable here.

I’m probably missing something but I’m not able to find what and where. Is there someone able to point me to the right direction?
Below is my configuration. The sofpthone is registered as 1060.

Thanks in advance. Marco Signorini.

pjsip.conf:

[transport-tls]
type=transport protocol=tls bind=0.0.0.0
cert_file=/etc/asterisk/sslcert.pem method=tlsv1

[1060]
type=endpoint transport=transport-tls context=from-internal use_avpf=yes media_encryption=sdes disallow=all allow=alaw allow=ulaw aors60
auth60

[1060]
type=auth auth_type=userpass password60
username60

[1060]
type=aor max_contacts

[204]
….

http.conf:

enabled=yes bindaddr.10.5.49
bindport