vicidial.org

Hi All,

I am struggling to take inbound calls in Vicidial using Patton PRI gateway

| Patton Model: SN4970/1E15V | Asterisk Version: 13.21.1 | With Opensuse 32.0 |

Below is the entery in ADMIN > CARRIER

Registration String: register => patton:123456@IPofPattonPRI:5060/patton

[pattoncarrier]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
type=friend
type=peer
svrlookup=yes
port=5060
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
insecure=invite
insecure=port,invite
fromuser=patton
nat=yes
host=IPofPattonPRI
username=patton
secret=123456

Globals String: pattoncarrier=sip/pattoncarrier

Dialplan Entry:
exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91X.,2,Set(CALLERID(num)=MyPRINumber)
exten => _91X.,3,Dial(SIP/${EXTEN:1}@metronet,,tTorR)
exten => _91X.,4,Hangup

===============================================================================
Continue receiving below error in Asterisk -rvvvv

[Oct 25 00:37:25] NOTICE[58482]: chan_sip.c:15906 sip_reg_timeout: -- Registration for 'patton@10.201.18.6' timed out, trying again (Attempt #81)
[Oct 25 00:37:25] WARNING[58482]: chan_sip.c:25083 handle_response: Host '10.201.18.6:5060' does not implement 'REGISTER'

===============================================================================
For Inbound calling below are the logs in Asterisk -rvvvv

[Oct 25 01:06:24] NOTICE[63010]: chan_sip.c:15906 sip_reg_timeout: -- Registration for 'patton@10.201.18.6' timed out, trying again (Attempt #65)
[Oct 25 01:06:24] WARNING[63010]: chan_sip.c:25083 handle_response: Host '10.201.18.6:5060' does not implement 'REGISTER'
[Oct 25 01:06:38] == Using SIP RTP CoS mark 5
[Oct 25 01:06:38] > 0x7fad68016da0 -- Strict RTP learning after remote address set to: 10.201.18.6:4918
[Oct 25 01:06:38] -- Executing [10000@trunkinbound:1] AGI("SIP/pattoncarrier-00000003", "agi-DID_route.agi") in new stack
[Oct 25 01:06:38] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Oct 25 01:06:38] -- <SIP/pattoncarrier-00000003>AGI Script agi-DID_route.agi completed, returning 0
[Oct 25 01:06:38] -- Executing [9998811112@default:1] Wait("SIP/pattoncarrier-00000003", "2") in new stack
[Oct 25 01:06:40] -- Executing [9998811112@default:2] Answer("SIP/pattoncarrier-00000003", "") in new stack
[Oct 25 01:06:40] > 0x7fad68016da0 -- Strict RTP switching to RTP target address 10.201.18.6:4918 as source
[Oct 25 01:06:40] -- Executing [9998811112@default:3] Playback("SIP/pattoncarrier-00000003", "ss-noservice") in new stack
[Oct 25 01:06:40] -- <SIP/pattoncarrier-00000003> Playing 'ss-noservice.gsm' (language 'en')
[Oct 25 01:06:43] > 0x7fad68016da0 -- Strict RTP learning complete - Locking on source address 10.201.18.6:4918
[Oct 25 01:06:44] NOTICE[63010]: chan_sip.c:15906 sip_reg_timeout: -- Registration for 'patton@10.201.18.6' timed out, trying again (Attempt #66)
[Oct 25 01:06:44] WARNING[63010]: chan_sip.c:25083 handle_response: Host '10.201.18.6:5060' does not implement 'REGISTER'
[Oct 25 01:06:45] -- Executing [9998811112@default:4] Playback("SIP/pattoncarrier-00000003", "vm-goodbye") in new stack
[Oct 25 01:06:45] -- <SIP/pattoncarrier-00000003> Playing 'vm-goodbye.gsm' (language 'en')
[Oct 25 01:06:46] -- Executing [9998811112@default:5] Hangup("SIP/pattoncarrier-00000003", "") in new stack
[Oct 25 01:06:46] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/pattoncarrier-00000003'
[Oct 25 01:06:46] WARNING[601][C-00000003]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Oct 25 01:06:46] -- Executing [h@default:1] AGI("SIP/pattoncarrier-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Oct 25 01:06:46] -- <SIP/pattoncarrier-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
===============================================================================

** Above logs of Asterisk -rvvv are showing call is landing on "VM-goodbye"

** I am also receiving Vicidial voice mail (VM-goodbye).

But call is not landing on agent, please anyone help me to resolve the issue.

Step one: Get it working in Asterisk. If it's designed to work with Asterisk, the manufacturer and/or their support board will have instructions to make it work with vanilla asterisk. After that's done, and you have successfully generated a call with sound in both directions that terminates upon request properly, we can show you how to make minor alterations for it to work with vicidial.

Start here: Host '10.201.18.6:5060' does not implement 'REGISTER'

What makes you think you need to register? Is there a management login for the PRI that provides you with this information and the user/pass?

WHY are you using a SIP PRI? You'll be locked to a single carrier who is also your ISP, this tends to lead to excessive invoicing and a multi-year contract. Often this turns into a very bad situation.

Hi williamconley,

Thanks for the reply.

Please find my responses and query below -

When I install vicidial with asterisk, then why need to work directly in Asterisk??? .... I am learning

Providers provided us this PRI Line which is a telecommunication standard that enables traditional phone lines to carry voice and data.

We have connected this PRI line in our PRI gateway (PATTON). In the PATTON we have created a SIP account (User/Pass). Which we are using in Vicidial peer [pattoncarrier]

in Admin > Carrier. I don't know how I can communicate my PRI Gateway (PATTON) with Vicidial?

PRI Line which is a telecommunication standard that enables traditional phone lines to carry voice and data.

PRI splits between internet and Voice. And if your ISP uses a version that allows SIP for the voice, then it's actually ALL internet, but you're paying THEM for landlines while using them as a VOIP carrier. But since they own the equipment, and they are your ISP, they will treat you verify differently than other VOIP carriers. They will treat you like a client with no other options and often charge you double (I've seen up to 10 times) the price of another VOIP carrier for the same service. Howver: That PRI could be use for ALL internet directly to you with no voice and you could use ANY other VOIP carrier and receive the exact same service from another carrier, but then you'd have a Set Fee for monthly internet from the ISP. Usually this results in a much lower total.

But we'll talk again after you've been paying a few months. In the end it boils down to the total cost per minute over the course of the service INCLUDING the monthly internet costs. Then compare it with 1c per minute (usually attainable, at least in the US) plus internet costs. If you get lucky you won't get ripped off.

The reason you have to deal with Asterisk directly is because you have a non-standard VOIP carrier. The box is rejecting your attempts to register. Instead of opening a ticket at a standard VOIP Carrier, you'll have to work with your ISP and/or the equipment manual/manufacturer. If you were using a generic carrier, you'd likely have no need of this.

The patton box is already rejecting your registration attempt. Doesn't matter what you put in admin->carriers until you resolve that issue if registration is a requirement.