[asterisk-users] SIP Connection Question
Hi All, I have a question about how a particular situation would work between two PBX systems: If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same rack, same network), and then pass a call from the Mitel to Asterisk to perform some functions (lookups, maybe routing), and then pass the call back to the Mitel to be routed to it's endpoint, would Asterisk stay in that loop after the call was passed back to the Mitel? Or, does the call leave Asterisk completely when passed back? If it does leave/stay in the loop, is there a way to force it to leave/stay based on what my needs are? Thanks, Kenny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Question
Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodríguez -Original Message- From: Kenneth Noisewater noisewater...@gmail.com Date: Thu, 1 Apr 2010 16:50:47 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Connection Question -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Question
On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote: Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodríguez -Original Message- From: Kenneth Noisewaternoisewater...@gmail.com Date: Thu, 1 Apr 2010 16:50:47 To:asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Connection Question OK, so for instance if I passed a call to Asterisk and grabbed CID info and did some lookups and then transferred it back to mitel to route to a user, then * would be out of the call path (loop, whatever). But, if I were to answer that call in * with an IVR to collect caller input to use and then transferred the call back to the Mitel to route to the endpoint, * would remain in the call. Is that a correct understanding? Also one more question, and please excuse my ignorance (I'm just a developer with pretty limited knowledge on the telephony side of things): When I talk about connecting the Mitel box and the Asterisk box together via a SIP trunk, is that trunk equal to 1 analog line, or channel or whatever, or can I make as many connections as I want on that trunk? Again, my knowledge is a bit limited, and thusfar people have been using a lot of terms interchangably with me to add to my confusion :). This only concerns me because I'm pretty sure we have to buy a license for each SIP trunk with Mitel. It would be really great if I could work out a solution like this, it will allow me to prove Asterisk's worth to my management, and open up a lot of doors for us and our internal apps. The Mitel SDK is unfortunately rather limited, but management is not in any way interested in jumping ship from Mitel to Asterisk. Personally, I say jump, I've had great experience with Asterisk, even in fairly heavy use situations. Anyone have any input on selling Asterisk to higher up's? I know there is the whole enterprise support aspect, but my team manages the Mitel stuff as it is anyway, and I think we'd all much rather be dealing with Asterisk/SER as the core solution. Thanks everyone for your input! Kenny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Question
If * answers the call, it will be on the loop but with canreinvite or directrtp the media can be out of * and redirected to the final end point even if signaling goes through *. For the trunk, you can have multiple simultaneous calls. I do not know about Mitel's licensing but with only one trunk you can have as much calls as * supports. Saludos, Juan E. Rodríguez -Original Message- From: Dr. Kenneth Noisewater noisewater...@gmail.com Date: Thu, 01 Apr 2010 19:35:54 To: jerdg...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP Connection Question On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote: Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodríguez -Original Message- From: Kenneth Noisewaternoisewater...@gmail.com Date: Thu, 1 Apr 2010 16:50:47 To:asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Connection Question OK, so for instance if I passed a call to Asterisk and grabbed CID info and did some lookups and then transferred it back to mitel to route to a user, then * would be out of the call path (loop, whatever). But, if I were to answer that call in * with an IVR to collect caller input to use and then transferred the call back to the Mitel to route to the endpoint, * would remain in the call. Is that a correct understanding? Also one more question, and please excuse my ignorance (I'm just a developer with pretty limited knowledge on the telephony side of things): When I talk about connecting the Mitel box and the Asterisk box together via a SIP trunk, is that trunk equal to 1 analog line, or channel or whatever, or can I make as many connections as I want on that trunk? Again, my knowledge is a bit limited, and thusfar people have been using a lot of terms interchangably with me to add to my confusion :). This only concerns me because I'm pretty sure we have to buy a license for each SIP trunk with Mitel. It would be really great if I could work out a solution like this, it will allow me to prove Asterisk's worth to my management, and open up a lot of doors for us and our internal apps. The Mitel SDK is unfortunately rather limited, but management is not in any way interested in jumping ship from Mitel to Asterisk. Personally, I say jump, I've had great experience with Asterisk, even in fairly heavy use situations. Anyone have any input on selling Asterisk to higher up's? I know there is the whole enterprise support aspect, but my team manages the Mitel stuff as it is anyway, and I think we'd all much rather be dealing with Asterisk/SER as the core solution. Thanks everyone for your input! Kenny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Problems
I had issues using NAT when having multiple phones as well as single phones behind NAT. You can try setting port forwarding on the phones side as well as look at a better router. Some routers will make you pull your hair out while others will work almost perfectly (this explains my now bald head :) ) Dovid - Original Message - From: Barry Fawthrop [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 12, 2006 7:27 AM Subject: [asterisk-users] SIP Connection Problems Hi All I have a Cisco 7960 which is connected remotely to an Asterisk server. Both are unfortunately behind NAT. The Phone registers and is show in sip show peers, with the correct public ip for the phone and a 100ms qualify time (1) I can dial the phone from another phone, it will ring but no voice goes through in fact I get this error on * console SIP response 481 Call Leg/Transaction Does Not Exist (2) the phone can make calls outbound fine, with voice no problems ? http://channels.debian.net/paste/3409 holds the SIP debug for the phone extension 650 port forwarding is also set on both sides, and sip.conf has the nat=yes , externip and localnet all set correctly Thank always Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Problems
Thanks Dovid I have port forwarding enabled on the linksys router ports 5060 and 1-2. I was wondering if I should also enable DMZ to the internal IP address of the phone ? Thanks Barry Dovid Bender wrote: I had issues using NAT when having multiple phones as well as single phones behind NAT. You can try setting port forwarding on the phones side as well as look at a better router. Some routers will make you pull your hair out while others will work almost perfectly (this explains my now bald head :) ) Dovid - Original Message - From: Barry Fawthrop [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 12, 2006 7:27 AM Subject: [asterisk-users] SIP Connection Problems ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Problems
Yes. It should help. See what happens. Also it can be your router at either or both ends. - Original Message - From: Barry Fawthrop [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 13, 2006 9:23 AM Subject: Re: [asterisk-users] SIP Connection Problems Thanks Dovid I have port forwarding enabled on the linksys router ports 5060 and 1-2. I was wondering if I should also enable DMZ to the internal IP address of the phone ? Thanks Barry Dovid Bender wrote: I had issues using NAT when having multiple phones as well as single phones behind NAT. You can try setting port forwarding on the phones side as well as look at a better router. Some routers will make you pull your hair out while others will work almost perfectly (this explains my now bald head :) ) Dovid - Original Message - From: Barry Fawthrop [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 12, 2006 7:27 AM Subject: [asterisk-users] SIP Connection Problems ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Connection Problems
Hi All I have a Cisco 7960 which is connected remotely to an Asterisk server. Both are unfortunately behind NAT. The Phone registers and is show in sip show peers, with the correct public ip for the phone and a 100ms qualify time (1) I can dial the phone from another phone, it will ring but no voice goes through in fact I get this error on * console SIP response 481 Call Leg/Transaction Does Not Exist (2) the phone can make calls outbound fine, with voice no problems ? http://channels.debian.net/paste/3409 holds the SIP debug for the phone extension 650 port forwarding is also set on both sides, and sip.conf has the nat=yes , externip and localnet all set correctly Thank always Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Connection Problems
Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 1-2). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Connection Problems
5000-600? Do you mean 5060? That is the port for 5060. 1-2 is for RTP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk UsersSent: Sunday, September 11, 2005 12:46 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Connection Problems Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 1-2). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Connection Problems
Are you using the Linksys router as your PPPoE termination or are using the Netopia?? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk Users Sent: Sunday, September 11, 2005 3:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Connection Problems Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 1-2). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP connection
I need help to make a conection form FWD to my pbx, I can receive a call from PSTN for a FXo card but know I need to receive call via IP form FWD I have activate hte IAX on freeworlddialup but does not work I can't make or receive calls. I virtually new in this can please somebody help me. thanks, scorpionny ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Connection Timing Out BroadVoice
I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI sip show registry HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 3584 Registered *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status sip.broadvoice.com/609299 147.135.0.128 255.255.255.255 5060 Unmonitored 1 sip peers [1 online , 0 offline] Any ideas why this is happening? Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Connection Timing Out BroadVoice
I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI sip show registry HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 3584 Registered *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status sip.broadvoice.com/609299 147.135.0.128 255.255.255.255 5060 Unmonitored 1 sip peers [1 online , 0 offline] Any ideas why this is happening? Try adding qualify=2000 to the sip.conf definition for broadvoice. See if that helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Connection Timing Out BroadVoice
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI sip show registry HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 3584 Registered *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status sip.broadvoice.com/609299 147.135.0.128 255.255.255.255 5060 Unmonitored 1 sip peers [1 online , 0 offline] Any ideas why this is happening? Try adding qualify=2000 to the sip.conf definition for broadvoice. See if that helps. Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Connection Timing Out BroadVoice
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote: I just signed up and configured a SIP connection from BroadVoice. It works great. This issue I have is that it seems after a couple calls (or a certain amount of time) Asterisk doesn't seem to be receiving these calls anymore. It seems as if BroadVoice is not redirecting the call to my Asterisk. Asterisk still seems to be ready for the call: *CLI sip show registry HostUsername Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 3584 Registered *CLI sip show peers Name/username HostDyn Nat ACL Mask Port Status sip.broadvoice.com/609299 147.135.0.128 255.255.255.255 5060 Unmonitored 1 sip peers [1 online , 0 offline] Any ideas why this is happening? Try adding qualify=2000 to the sip.conf definition for broadvoice. See if that helps. I think that worked thanks! BROADVOICE: Maybe you could add that option to the configuration at : http://www.broadvoice.com/support_install_asterisk.html for new customers. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Connection to a Cisco AS5xxx gateway
Hello, Does anybody have any experience connecting Asterisk to a Cisco gateway? I'm trying to terminate calls into this gateway, and then route incoming DID numbers from the gateway into Asterisk. So far, Asterisk sends the call to the gateway, and it connects the call, but there's no audio. I'm using the Cisco gateway with IOS 12.3.10T, connecting as SIP, no registration, and as clients I tried different SIP Phones including Cisco ATA (which connects to the gateway just fine without using asterisk), Gandstream ATA and the console. They all communicate to each other through SIP, but not to the Cisco gateway. I'm using g.711uLaw as the codec to talk to the gateway. Thanks, E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Connection to a Cisco AS5xxx gateway
Here is what works for me. It is currently working and in service on an MC3810.
plar is needed so incoming calls ring an extension in asterisk. extension 102 sends
call to my IVR root. Please remember to configure default gateway. This especially
important if you have nat specified in asterisk.
This is for an MC3810, but you should be able to get enough out of it to make your
AS5300 work.
Jojo
In IOS:
version 12.3
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname MC3810-1
!
boot-start-marker
boot system flash:mc3810-a2isv5-mz.123-10.bin
boot-end-marker
!
enable password 7 xxx
!
network-clock base-rate 56k
no aaa new-model
ip subnet-zero
!
no ip domain lookup
!
voice class codec 10
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 4 g729r8
codec preference 6 g729ar8
!
no voice confirmation-tone
!
controller T1 0
shutdown
framing sf
linecode ami
!
interface Ethernet0
ip address 192.168.1.7 255.255.255.0
ip route-cache same-interface
!
interface Serial0
no ip address
shutdown
!
interface Serial1
no ip address
shutdown
!
interface FR-ATM20
no ip address
shutdown
!
ip default-gateway 192.168.1.1
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.1.1
no ip http server
!
!
!
!
voice-port 1/2
connection plar 102
station-id name FXO2
station-id number 8002
!
voice-port 1/3
connection plar 102
station-id name FXO3
station-id number 8003
!
dial-peer cor custom
!
dial-peer voice 1 pots
destination-pattern ...
port 1/3
!
dial-peer voice 2 pots
destination-pattern ...
port 1/2
!
dial-peer voice 10 voip
destination-pattern 102
voice-class codec 10
session protocol sipv2
session target sip-server
!
sip-ua
retry invite 3
retry cancel 2
sip-server ipv4:192.168.1.5:5060
!
!
line con 0
exec-timeout 0 0
logging synchronous
transport preferred all
transport output all
line aux 0
transport preferred all
transport output all
line 2 3
transport preferred all
transport output all
line vty 0 4
password 7 x
login
transport preferred all
transport input all
transport output all
!
end
In sip.conf:
[8002]
type=friend
username=8002
host=192.168.1.7 - IP address of Cisco
canreinvite=no
qualify=yes
nat=no
dtmfmode=inband
[8003]
type=friend
username=8003
host=192.168.1.7 - IP address of Cisco
canreinvite=no
qualify=yes
nat=no
dtmfmode=inband
In extensions.conf
[default]
include = 8002
exten = 102,1,Goto(locals,s,1) -sends to root of my IVR
[8002]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
From: [EMAIL PROTECTED] on behalf of Emilio Panighetti
Sent: Tue 10/12/2004 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Connection to a Cisco AS5xxx gateway
Hello,
Does anybody have any experience connecting Asterisk to a Cisco gateway?
I'm trying to terminate calls into this gateway, and then route
incoming DID numbers from the gateway into Asterisk.
So far, Asterisk sends the call to the gateway, and it connects the
call, but there's no audio. I'm using the Cisco gateway with IOS
12.3.10T, connecting as SIP, no registration, and as clients I tried
different SIP Phones including Cisco ATA (which connects to the gateway
just fine without using asterisk), Gandstream ATA and the console. They
all communicate to each other through SIP, but not to the Cisco
gateway. I'm using g.711uLaw as the codec to talk to the gateway.
Thanks,
E.
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Re: [Asterisk-Users] SIP Connection to a Cisco AS5xxx gateway
Thanks. I looked at your config and it looks very simple (I would use dial-peers instead of plar, but that's not the issue). It seems that the gateway is sending the codec information mime-encoded, which is what I see from an ethereal trace. On the other hand, Asterisk doesn't seem to understand that. The Cisco gateway is sending RPT packets, but apparently, Asterisk is ignoring them. Asterisk, in debug mode, shows me the following errors: -- Executing Dial(SIP/1234-a56a, SIP/PSTN/4567) in new stack -- Called PSTN/4567 Oct 12 18:46:09 NOTICE[1107327920]: chan_sip.c:2617 process_sdp: Content is 'multipart/mixed;boundary=uniqueBoundary', not 'application/sdp' -- SIP/PSTN-934f answered SIP/1234-a56a -- Attempting native bridge of SIP/1234-a56a and SIP/PSTN-934f Oct 12 18:46:09 WARNING[1109470128]: rtp.c:460 ast_rtp_read: RTP Read too short (20, expecting 77524) RFC3389: 1 bytes, level 8... Oct 12 18:46:12 NOTICE[1109470128]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible == Spawn extension (from-sip, 4567, 1) exited non-zero on 'SIP/1234-a56a' Your gateway seems to have a comparable IOS. I don't think Cisco would change the SIP stack from platform to plaform. I'm using a CVS version of asterisk (Asterisk CVS-HEAD-10/11/04-01:13:29). Should I go back and use a 1.0 release? Everything else seems to be working so far. On Oct 12, 2004, at 5:36 PM, Asterisk wrote: Here is what works for me. It is currently working and in service on an MC3810. plar is needed so incoming calls ring an extension in asterisk. extension 102 sends call to my IVR root. Please remember to configure default gateway. This especially important if you have nat specified in asterisk. This is for an MC3810, but you should be able to get enough out of it to make your AS5300 work. Jojo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
