Comments on: Cisco router with FXO as an Asterisk gateway

By: Jim

Jim — Mon, 07 Feb 2011 16:32:32 +0000

Hy all, I have made the interconnection wint 1760 and Asterisk, and everything is OK (For about 15 Minutes). After that I get in the CLI:
===============================
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.0.56, port 5060
Audio is at 172.16.12.59 port 18248
Adding codec 0×8 (alaw) to SDP
Adding non-codec 0×1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.56:55350:
INVITE sip:2310688184@192.168.0.56:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.12.59:5060;branch=z9hG4bK69033545;rport
Max-Forwards: 70
From: ;tag=as1fc2060e
To: ;tag=E8764B79-16C6
Contact:
Call-ID: FF2BA038-319711E0-9C32BD05-1C07ED8A@192.168.0.56
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.6.2.13)
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1270771100 1270771100 IN IP4 172.16.12.59
s=Asterisk PBX 1.6.2.13
c=IN IP4 172.16.12.59
t=0 0
m=audio 18248 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

—

SIP/2.0 420 Bad Extension
Via: SIP/2.0/UDP 172.16.12.59:5060;branch=z9hG4bK69033545;rport
From: ;tag=as1fc2060e
To: ;tag=E8764B79-16C6
Call-ID: FF2BA038-319711E0-9C32BD05-1C07ED8A@192.168.0.56
CSeq: 102 INVITE
Unsupported: timer
Content-Length: 0

— (8 headers 0 lines) —
— Got SIP response 420 “Bad Extension” back from 192.168.0.56
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.0.56, port 5060
Transmitting (NAT) to 192.168.0.56:5060:
ACK sip:2310688184@192.168.0.56:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.12.59:5060;branch=z9hG4bK69033545;rport
Max-Forwards: 70
From: ;tag=as1fc2060e
To: ;tag=E8764B79-16C6
Contact:
Call-ID: FF2BA038-319711E0-9C32BD05-1C07ED8A@192.168.0.56
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.6.2.13)
Content-Length: 0

—
— Executing [h@macro-dial:1] Macro(“SIP/Cisco-0000009f”, “hangupcall”) in new stack
— Executing [s@macro-hangupcall:1] GotoIf(“SIP/Cisco-0000009f”, “1?skiprg”) in new stack
— Goto (macro-hangupcall,s,4)
— Executing [s@macro-hangupcall:4] GotoIf(“SIP/Cisco-0000009f”, “0?skipblkvm”) in new stack
— Executing [s@macro-hangupcall:5] NoOp(“SIP/Cisco-0000009f”, “Cleaning Up Block VM Flag: BLKVM/600/SIP/Cisco-0000009f”) in new stack
— Executing [s@macro-hangupcall:6] NoOp(“SIP/Cisco-0000009f”, “Deleting: BLKVM/600/SIP/Cisco-0000009f “) in new stack
— Executing [s@macro-hangupcall:7] GotoIf(“SIP/Cisco-0000009f”, “1?theend”) in new stack
— Goto (macro-hangupcall,s,9)
— Executing [s@macro-hangupcall:9] Hangup(“SIP/Cisco-0000009f”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/Cisco-0000009f’ in macro ‘hangupcall’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/Cisco-0000009f’ in macro ‘dial’
== Spawn extension (ext-group, 600, 14) exited non-zero on ‘SIP/Cisco-0000009f’
== MixMonitor close filestream
== End MixMonitor Recording SIP/Cisco-0000009f
Elastix203*CLI>
===============================

Have you ever met the same eroor anyone?

By: adam

adam — Sat, 08 Jan 2011 22:34:47 +0000

Hey i know this thread is old and i believe it got these all these hits because it really was ahead of its time…… I am currently in the process of implementing a Cisco 3825 as a gateway for 3CX……. has anybody attempted this and if so could you shed any light on the matter……… Thanks a Million Adam……..
akusta@itlynk.com

By: Matt Miller

Matt Miller — Tue, 08 Sep 2009 04:58:30 +0000

You’re correct on my setup. I had played with the ephone-hunt command, but didn’t think it would work given that the fxs ports aren’t ephones. I did discover the voice hunt-group command that in theory will do what I need. I haven’t had any luck confugiring it from home. I’ll have to drive over to that office and play with it when I get some time. From what I could see from a debug voice dialpeer detail, it’s trying to match an incoming and outgoing dialpeer and I never get transferred to Asterisk. Asterisk never even shows an inbound SIP connection. I tried a sequential hunt group, but there are several other types available now, so I’ll have to mess with those. If I get something to work, I’ll post the configs in case someone else is looking to do something similar.

By: scott

scott — Tue, 08 Sep 2009 02:28:44 +0000

Matt, I’m not real clear on what your setup is. Sounds like you have PSTN lines on the FXO’s and other SIP trunks from offices. You have analog phones at that office on the FXS ports (not IP phones). You want the Asterisk to only provide VM and get out of the way for regular calls if it’s down.

Best bet is probably the hunt group like you suggested. I don’t have time to build that right now but basically the first hop in the hunt group is the FXS extension and the second is the Asterisk trunk to the VM. If it gets to Asterisk it immediately goes to VM. Just be sure to work through the hunt group logic to be sure you aren’t black-holing the call if Asterisk is down.

By: Matt Miller

Matt Miller — Tue, 08 Sep 2009 01:37:39 +0000

@Matt Miller I should also add that the 1760 is not part of a larger call manager configuration. All of the toll bypass is done with static SIP connections between phone systems. I've seen mention to STCAPP online, but only in reference to call manager, not call manager express (which is what's on the 1760). It sounds like what I'm trying to do, but I don't know that it's supported.

By: Matt Miller

Matt Miller — Tue, 08 Sep 2009 01:24:31 +0000

Hi. I came across your page when trying to find a solution to my problem. I’ve got a 1760 (latest T-train IOS, max memory) with 4 FXO ports and 4 FXS ports. I’m currently using the 1760 to provide toll bypass to remote offices and I’d like to use it to provide voicemail for an analog system whose voicemail component died (I’d prefer to use Asterisk anyway, but the old one going is forcing me to push this to the top of my priority list).

The FXO ports are configured with:

connection plar opx 18005551212

There are voice dial peers configured with 18005551212 as the destination pattern, so when a call comes in, it rings the first available FXS port (and remote offices can place a sip call to the 1760′s IP with the office’s main number as the destination). There are also dial peers configured for outbound calls to the analog lines as well as our remote locations.

Everything to this point has been working flawlessly. I’ve setup a dialpeer to asterisk, so when I dial 1191 from one of the analog ports, it routes to asterisk. Again, this works. The problem I have is that I’m not sure how to configure call forward no answer for the FXO ports on the router. I know I could do it if I had 2 ATAs to provide the FXO ports, but I don’t want to invest in any more hardware.

I’ve got a fix, but I’m looking for a better solution. Right now, I can use connection plar opx to send all incoming calls to 1191 (the sip connection to asterisk), then have asterisk place a sip call back to the 1760 to ring the FXS ports. Asterisk would have the timeout built in to the dial command to divert the call to voicemail. I’d like to avoid this scenario because the Asterisk server is on older hardware and I do not want all incoming calls to be dependant on the Asterisk server.
If the server were to fail, there is nobody at that location that could troubleshoot/repair the server and if it goes out, it would prevent all incoming calls from ringing. I’m hoping that someone on here might have an idea on how I can get this to work. Maybe a hunt group, or something… I’m not sure how to do this since the dial-peer doesn’t have the CFNA options that ephones have.

By: Frank_Bauer

Frank_Bauer — Thu, 27 Aug 2009 14:00:34 +0000

O.. I have narrowed it down to the “qualify=yes” settings on the asterisk trunk and extension settings,asterisk is sending sip options to the gateway every 10 seconds by default, if no response is recieved in 2 seconds by default, then asterisk considers the peers / extensions unreachable. I have played arround with the default 10 seconds, for example “qualify=3000″ which makes the sip connection last for 3000 seconds, but then it goes down again. I’m looking into etending the time out, so I’m guessing the next logical step is to extend the hold default hold timer of 2 seconds.

By: Frank_Bauer

Frank_Bauer — Tue, 25 Aug 2009 17:14:58 +0000

.. thanks will look into it and get back to you , leaving to go home from work now, talk to you soon .

By: scott

scott — Tue, 25 Aug 2009 17:05:55 +0000

That’s a tough one without more info. Could be related to the timers in the sip-ua. Debug is your friend! Don’t forget to open a debug console on Asterisk to see what it’s trying to do.

By: Frank_Bauer

Frank_Bauer — Tue, 25 Aug 2009 15:54:07 +0000

Damnit.. spoke too soon, ok now the registration on the Gateway is timing out randomly.. any ideas ?