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Shaoul Jacobson - TELLINK

2005-Feb-22 04:02 UTC

[Asterisk-Users] Call Manager Express Peer

Hi,

There seem to be some codec incompatibility.
On *, you define alaw and you set ulaw on the Cisco.

Set both to same or add the other codec on (at least) one side.
Try if that solve it

Ex:
Add "allow ulaw" on * after the "allow alaw"
And / or
Add "codec g711alaw" on Cisco above the "codec g711ulaw"

If I remember correctly, Cisco parse the codecs according to their entry
onder. Asterisk orders according to alphabetical order.

If you do not need both codecs, set only one to simplify.

Regards,


Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :	+32 3 201 96 36
Fax : 	+32 3 227 09 81
e-mail	shaoul@tellink.com


-----Original Message-----
From: Nathan Alberti [mailto:na@nathanalberti.com] 
Sent: mercredi 23 f?vrier 2005 0:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call Manager Express Peer


I have the following configuration and am obviously missing something 
small that is causing * not to work as expected.


I have the following defined in sip.conf

[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes

and [devel_in] is defined in extentions.conf

However when I try to call via the dial peer I have configured on the 
cisco (below) I get :

Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'

Which is correct, meaning the context declaration is not being respected.

------
dial-peer voice 101 voip
 destination-pattern 10.
 session protocol sipv2
 session target ipv4:10.0.0.133
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
-------


My bad or something else ??

TIA,

Nathan.



Here is a sip debug for that peer:


Sending to 10.0.9.1 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.0.9.1:19206
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for 101 in default
Feb 22 18:44:25 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.9.1:5060;branch=z9hG4bK3047A
From: "Test Phone 1" <sip:95555001@10.0.9.1>;tag=17AFD44-10AD
To: <sip:101@10.0.0.133>;tag=as3edc130d
Call-ID: 2C2C3A74-83F511D9-8450EFE0-1F555CD9@10.0.9.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101@10.0.0.133>
Content-Length: 0


 to 10.0.9.1:5060
Destroying call '2C2C3A74-83F511D9-8450EFE0-1F555CD9@10.0.9.1'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.9.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.133:5060;branch=z9hG4bK2b06e290
From: "asterisk" <sip:asterisk@10.0.0.133>;tag=as0a8b5343
To: <sip:10.0.9.1>
Contact: <sip:asterisk@10.0.0.133>
Call-ID: 6d840c056f0f06c241e744263a64623b@10.0.0.133
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 22 Feb 2005 10:44:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 10.0.9.1:5060
Destroying call '6d840c056f0f06c241e744263a64623b@10.0.0.133'



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Robert Hajime Lanning

2005-Feb-22 09:03 UTC

head link

[Asterisk-Users] Call Manager Express Peer

<quote who="Nathan Alberti">> I have the following defined in sip.conf
>
> [ccme-in]
> type=peer
> host=10.0.9.1
> context=devel_in
> disallow=all
> allow=alaw
> nat=no
> canreinvite=yes
> qualify=yes
>
> and [devel_in] is defined in extentions.conf
>
> However when I try to call via the dial peer I have configured on the
> cisco (below) I get :
type=peer is for going out of asterisk to the "peer"
For inbound calls, type=user
For one entry that does both, type=friend

Though, it is recommended to have two entries, one peer for outbound
calls and one user for inbound.

-- 
END OF LINE
       -MCP

Greg Oliver

2005-Feb-22 15:40 UTC

head link

[Asterisk-Users] Call Manager Express Peer

The only thing I have different in my CME dial-peers is "application 
session" for each of them.  Other than that, what you have works for me..

-Greg

Nathan Alberti wrote:> 
> I have the following configuration and am obviously missing something 
> small that is causing * not to work as expected.
> 
> 
> I have the following defined in sip.conf
> 
> [ccme-in]
> type=peer
> host=10.0.9.1
> context=devel_in
> disallow=all
> allow=alaw
> nat=no
> canreinvite=yes
> qualify=yes
> 
> and [devel_in] is defined in extentions.conf
> 
> However when I try to call via the dial peer I have configured on the 
> cisco (below) I get :
> 
> Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
> find extension context 'default'
> 
> Which is correct, meaning the context declaration is not being respected.
> 
> ------
> dial-peer voice 101 voip
> destination-pattern 10.
> session protocol sipv2
> session target ipv4:10.0.0.133
> dtmf-relay rtp-nte
> codec g711ulaw
> no vad
> -------
> 
> 
> My bad or something else ??
> 
> TIA,
> 
> Nathan.
> 
> 
> 
> Here is a sip debug for that peer:
> 
> 
> Sending to 10.0.9.1 : 5060 (non-NAT)
> Found RTP audio format 0
> Found RTP audio format 101
> Peer audio RTP is at port 10.0.9.1:19206
> Found description format PCMU
> Found description format telephone-event
> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 
> (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
> 0x1 (g723)
> Looking for 101 in default
> Feb 22 18:44:25 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
> find extension context 'default'
> Reliably Transmitting (no NAT):
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 10.0.9.1:5060;branch=z9hG4bK3047A
> From: "Test Phone 1"
<sip:95555001@10.0.9.1>;tag=17AFD44-10AD
> To: <sip:101@10.0.0.133>;tag=as3edc130d
> Call-ID: 2C2C3A74-83F511D9-8450EFE0-1F555CD9@10.0.9.1
> CSeq: 101 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:101@10.0.0.133>
> Content-Length: 0
> 
> 
> to 10.0.9.1:5060
> Destroying call '2C2C3A74-83F511D9-8450EFE0-1F555CD9@10.0.9.1'
> 11 headers, 0 lines
> Reliably Transmitting:
> OPTIONS sip:10.0.9.1 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.133:5060;branch=z9hG4bK2b06e290
> From: "asterisk" <sip:asterisk@10.0.0.133>;tag=as0a8b5343
> To: <sip:10.0.9.1>
> Contact: <sip:asterisk@10.0.0.133>
> Call-ID: 6d840c056f0f06c241e744263a64623b@10.0.0.133
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Date: Tue, 22 Feb 2005 10:44:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Length: 0
> 
> (no NAT) to 10.0.9.1:5060
> Destroying call '6d840c056f0f06c241e744263a64623b@10.0.0.133'
> 
> 
> 
> _______________________________________________
> 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
> 
>

Nathan Alberti

2005-Feb-22 16:44 UTC

head link

[Asterisk-Users] Call Manager Express Peer

I have the following configuration and am obviously missing something 
small that is causing * not to work as expected.


I have the following defined in sip.conf

[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes

and [devel_in] is defined in extentions.conf

However when I try to call via the dial peer I have configured on the 
cisco (below) I get :

Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'

Which is correct, meaning the context declaration is not being respected.

------
dial-peer voice 101 voip
 destination-pattern 10.
 session protocol sipv2
 session target ipv4:10.0.0.133
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
-------


My bad or something else ??

TIA,

Nathan.



Here is a sip debug for that peer:


Sending to 10.0.9.1 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.0.9.1:19206
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for 101 in default
Feb 22 18:44:25 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.9.1:5060;branch=z9hG4bK3047A
From: "Test Phone 1" <sip:95555001@10.0.9.1>;tag=17AFD44-10AD
To: <sip:101@10.0.0.133>;tag=as3edc130d
Call-ID: 2C2C3A74-83F511D9-8450EFE0-1F555CD9@10.0.9.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101@10.0.0.133>
Content-Length: 0


 to 10.0.9.1:5060
Destroying call '2C2C3A74-83F511D9-8450EFE0-1F555CD9@10.0.9.1'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.9.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.133:5060;branch=z9hG4bK2b06e290
From: "asterisk" <sip:asterisk@10.0.0.133>;tag=as0a8b5343
To: <sip:10.0.9.1>
Contact: <sip:asterisk@10.0.0.133>
Call-ID: 6d840c056f0f06c241e744263a64623b@10.0.0.133
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 22 Feb 2005 10:44:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 10.0.9.1:5060
Destroying call '6d840c056f0f06c241e744263a64623b@10.0.0.133'

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