On your pots dial-peer set the codec to g711alaw, or configure your SIP clients to use that codec.
Dan
-----Original Message----- From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de] Sent: Friday, April 25, 2003 7:55 AM To: serusers@lists.iptel.org Subject: [Serusers] problem with cisco 2600 to pstn
Hi all,
I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented
^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ------------------------------------------------------------------------ ---- ------------
Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law.
So if I call the router from a normal telephone, the debugging looks as follows: ------------------------------------------------------------------------ ---- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ------------------------------------------------------------------------ ---- ---------
whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law.
I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this.
Has anybody in this mailinglist the same experience?
Any hints would be very appreciated.
Thanks
yang
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
codec set at the voip dial-peer with session target to sip server
cmiiw
budi --- Dan Austin Dan_Austin@Phoenix.com wrote:
On your pots dial-peer set the codec to g711alaw, or configure your SIP clients to use that codec.
Dan
-----Original Message----- From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de] Sent: Friday, April 25, 2003 7:55 AM To: serusers@lists.iptel.org Subject: [Serusers] problem with cisco 2600 to pstn
Hi all,
I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says:
------------------------------------------------------------------------
00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2
^^^^^^00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented
^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available
------------------------------------------------------------------------
Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law.
So if I call the router from a normal telephone, the debugging looks as follows:
------------------------------------------------------------------------
01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3
^^^01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s
------------------------------------------------------------------------
whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law.
I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this.
Has anybody in this mailinglist the same experience?
Any hints would be very appreciated.
Thanks
yang
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Just FYI I though I'd post by BRI configuration Here as well. It might help somebody.
I've got BRI ports running on one of my routers. Here is my software version:
jabber#show ver Cisco Internetwork Operating System Software IOS (tm) 3600 Software (C3640-JS-M), Version 12.2(13b), RELEASE SOFTWARE (fc1)
You have to have VWICs, not normla BRIs, my hardware includes: FRU Part Number: NM-2V (this takes two VWIC cards) TE BRI Voice daughter card (2 port, 4 channels) (VWIC 2B-ST)
Here is my configuration:
! Put the codecs you want to use here: voice class codec 1 codec preference 1 g711ulaw !
! Configure the BRI SPIDS like this: interface BRI1/0 no ip address isdn switch-type basic-ni isdn spid1 97224277470101 2427747 isdn spid2 97224299170101 2429917 isdn incoming-voice voice isdn sending-complete ! voice-port 1/0/0 echo-cancel coverage 32 description 9722427747,9917 ! voice-port 1/0/1 echo-cancel coverage 32 description 9723234377,4378 ! ! You have to configure outgoing patterns for each channel. ! These are US long distance, eg, 14085551212, preceeded by a 9 ! I guess you could just use a pattern like T if you want everything ! To go out. The tricky part here is you have to recognise which direction ! The call is going. You have inbound calls, and outbound calls. ! I use the '9' to queue to myself that the call came from somewhere ! Else and is destined for POTS. If you are using this for inbound ! You need to set up at least one 'direct-inward-dial' for each pots ! Port... I believe in this example below we allow inbound calls ! From just port 0, channel 0. I don't actually know what would happen ! With an inbound call on port 0, channel 1 in this configuration.
dial-peer voice 1 pots destination-pattern 91.......... port 1/0/0 forward-digits 11 direct-inward-dial ! dial-peer voice 2 pots destination-pattern 91.......... port 1/0/1 forward-digits 11 ! dial-peer voice 1100 voip destination-pattern T voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay h245-alphanumeric no vad ! ! sip-ua max-forwards 15 retry invite 3 retry response 3 retry bye 6 retry cancel 3 timers trying 1000 sip-server ipv4:IP.OF.YOUR.SER.PROXY.SERVER !
---greg
-----Original Message----- From: serusers-admin@lists.iptel.org [mailto:serusers-admin@lists.iptel.org] On Behalf Of budi wibowo Sent: Friday, April 25, 2003 10:57 AM To: serusers@lists.iptel.org Subject: RE: [Serusers] problem with cisco 2600 to pstn
codec set at the voip dial-peer with session target to sip server
cmiiw
budi --- Dan Austin Dan_Austin@Phoenix.com wrote:
On your pots dial-peer set the codec to g711alaw, or configure your SIP clients to use that codec.
Dan
-----Original Message----- From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de] Sent: Friday, April 25, 2003 7:55 AM To: serusers@lists.iptel.org Subject: [Serusers] problem with cisco 2600 to pstn
Hi all,
I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says:
00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2
^^^^^^00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented
^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available
Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law.
So if I call the router from a normal telephone, the debugging looks as follows:
01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3
^^^01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s
whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law.
I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this.
Has anybody in this mailinglist the same experience?
Any hints would be very appreciated.
Thanks
yang
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
budi wibowo -
Dan Austin -
Greg Fausak